remove extraneous svn:executable properties

git-svn-id: https://origsvn.digium.com/svn/asterisk/tags/0.1.5@7221 65c4cc65-6c06-0410-ace0-fbb531ad65f3

BUGS

@@ -2,8 +2,10 @@
   these bugs are in asterisk, and sometimes they relate to the products
   that asterisk uses.
 
-* The MP3 decoder is completely broken
-
 * The translator API may introduce warble in the case of going in both
   directions, but I haven't verified that.  The trouble should only enter
   in the case of mismatched frame lengths.
+
+* In general Asterisk is a very new program, and there are liable to be
+  many bugs yet to be discovered, so if you think you've found one, please
+  be sure to report it.

CHANGES

@@ -0,0 +1,43 @@
+* Asterisk 0.1.5
+ -- Fix a small mistake in IAX
+ -- Fix the QuickNet driver to work with newer cards
+* Asterisk 0.1.4
+ -- Update VoFR some more
+ -- Fix the QuickNet driver to work with LineJack
+ -- Add ability to pass images for IAX.
+* Asterisk 0.1.3
+ -- Update VoFR for latest sangoma code
+ -- Update QuickNet Driver
+ -- Add text message handling
+ -- Fix transfers to use "default" if not in current context
+ -- Add call parking
+ -- Improve format/content negotiation
+ -- Added support for multiple languages
+ -- Bug fixes, as always...
+* Asterisk 0.1.2
+ -- Updated README file with a "Getting Started" section
+ -- Added sample sounds and configuration files.
+ -- Added LPC10 very low bandwidth (low quality) compression
+ -- Enhanced translation selection mechanism.
+ -- Enhanced IAX jitter buffer, improved reliability
+ -- Support echo cancelation on PhoneJack
+ -- Updated PhoneJack driver to std. Telephony interface
+ -- Added app_echo for evaluating VoIP latency
+ -- Added app_system to execute arbitrary programs
+ -- Updated sample configuration files
+ -- Added OSS channel driver (full duplex only)
+ -- Added IAX implementation
+ -- Fixed some deadlocks.
+ -- A whole bunch of bug fixes
+* Asterisk 0.1.1
+ -- Revised translator, fixed some general race conditions throughout *
+ -- Made dialer somewhat more aware of incompatible voice channels
+ -- Added Voice Modem driver and A/Open Modem Driver stub
+ -- Added MP3 decoder channel
+ -- Added Microsoft WAV49 support
+ -- Revised License -- Pure GPL, nothing else
+ -- Modified Copyright statement since code is still currently owned by author
+ -- Added RAW GSM headerless data format
+ -- Innumerable bug fixes
+* Asterisk 0.1.0
+ -- Initial Release

CREDITS

@@ -0,0 +1,52 @@
+=== HARDWARE DONORS === 
+
+* Special thanks to Adtran, Inc, for their donation of an Adtran Atlas,
+and TSU 120e to the project. (http://www.adtran.com)
+
+* Thanks to QuickNet Technologies for their donation of an Internet
+PhoneJack card to the project.  (http://www.quicknet.net)
+
+=== OTHER SOURCE CODE IN ASTERISK ===
+
+I did not implement the codecs in asterisk.  Here is the copyright on the
+GSM source:
+
+Copyright 1992, 1993, 1994 by Jutta Degener and Carsten Bormann,
+Technische Universitaet Berlin
+
+Any use of this software is permitted provided that this notice is not
+removed and that neither the authors nor the Technische Universitaet Berlin
+are deemed to have made any representations as to the suitability of this
+software for any purpose nor are held responsible for any defects of
+this software.  THERE IS ABSOLUTELY NO WARRANTY FOR THIS SOFTWARE.
+
+As a matter of courtesy, the authors request to be informed about uses
+this software has found, about bugs in this software, and about any
+improvements that may be of general interest.
+
+Berlin, 28.11.1994
+Jutta Degener
+Carsten Bormann
+
+And the copyright on the ADPCM source:
+
+Copyright 1992 by Stichting Mathematisch Centrum, Amsterdam, The
+Netherlands.
+
+                        All Rights Reserved
+
+Permission to use, copy, modify, and distribute this software and its
+documentation for any purpose and without fee is hereby granted,
+provided that the above copyright notice appear in all copies and that
+both that copyright notice and this permission notice appear in
+supporting documentation, and that the names of Stichting Mathematisch
+Centrum or CWI not be used in advertising or publicity pertaining to
+distribution of the software without specific, written prior permission.
+
+STICHTING MATHEMATISCH CENTRUM DISCLAIMS ALL WARRANTIES WITH REGARD TO
+THIS SOFTWARE, INCLUDING ALL IMPLIED WARRANTIES OF MERCHANTABILITY AND
+FITNESS, IN NO EVENT SHALL STICHTING MATHEMATISCH CENTRUM BE LIABLE
+FOR ANY SPECIAL, INDIRECT OR CONSEQUENTIAL DAMAGES OR ANY DAMAGES
+WHATSOEVER RESULTING FROM LOSS OF USE, DATA OR PROFITS, WHETHER IN AN
+ACTION OF CONTRACT, NEGLIGENCE OR OTHER TORTIOUS ACTION, ARISING OUT
+OF OR IN CONNECTION WITH THE USE OR PERFORMANCE OF THIS SOFTWARE.

LICENSE

@@ -1,22 +1,3 @@
-=================================================================
-Please note that Asterisk is distributed under the GNU General 
-Public License, with the following additional notices (some of
-these notices may be revised or relaxed in future releases):
-
-	a) All distributions or derivative works MUST not remove
-the name of Linux Support Services, LLC or Adtran, Inc. from 
-the product.
-
-	b) Some codecs included in the full Asterisk distribution
-*may* be covered by software patents.  
-
-	c) If You bring legal action against an organization or 
-individual for patent infringement regarding any Open Source 
-software (as defined by the Open Source Definition available at
-http://www.opensource.org/osd.html) then your right to use and/or
-distribute this software is immediately terminated.
-
-=================================================================
 
 		    GNU GENERAL PUBLIC LICENSE
 		       Version 2, June 1991

Makefile

@@ -3,7 +3,7 @@
 # 
 # Top level Makefile
 #
-# Copyright (C) 1999, Adtran Inc. and Linux Support Services, LLC
+# Copyright (C) 1999, Mark Spencer
 #
 # Mark Spencer <markster@linux-support.net>
 #
@@ -16,16 +16,31 @@
 
 MODULES_DIR=/usr/lib/asterisk/modules
 
+# Pentium Pro Optimize
+#PROC=i686
+# Pentium Optimize
+PROC=i586
+
 DEBUG=-g #-pg
 INCLUDE=-Iinclude -I../include
-CFLAGS=-Wall -Werror -O6 $(DEBUG) $(INCLUDE) -D_REENTRANT
-CFLAGS+=$(shell if $(CC) -march=i686 -S -o /dev/null -xc /dev/null >/dev/null 2>&1; then echo "-march=i686"; fi)
+CFLAGS=-pipe -Wall -Werror -Wmissing-prototypes -Wmissing-declarations -O6 $(DEBUG) $(INCLUDE) -D_REENTRANT
+CFLAGS+=$(shell if $(CC) -march=$(PROC) -S -o /dev/null -xc /dev/null >/dev/null 2>&1; then echo "-march=$(PROC)"; fi)
 SUBDIRS=channels pbx apps codecs formats
-LIBS=-ldl -lpthread #-lefence
-OBJS=io.o sched.o logger.o frame.o loader.o config.o channel.o translate.o file.o say.o pbx.o asterisk.o
+LIBS=-ldl -lpthread -lreadline # -lefence
+OBJS=io.o sched.o logger.o frame.o loader.o config.o channel.o translate.o file.o say.o pbx.o cli.o md5.o asterisk.o
 CC=gcc
 INSTALL=install
 
+_all: all
+	@echo " +--------- Asterisk Build Complete ---------+"  
+	@echo " + Asterisk has successfully been built, but +"  
+	@echo " + cannot be run before being installed by   +"  
+	@echo " + running:                                  +"  
+	@echo " +                                           +"
+	@echo " +               make install                +"  
+	@echo " +                                           +"
+	@echo " +-------------------------------------------+"  
+
 all: asterisk subdirs
 
 asterisk: $(OBJS)
@@ -38,8 +53,52 @@ clean:
 	for x in $(SUBDIRS); do $(MAKE) -C $$x clean || exit 1 ; done
 	rm -f *.o *.so asterisk
 
+datafiles: all
+	mkdir -p /var/lib/asterisk/sounds/digits
+	for x in sounds/digits/*; do \
+		install $$x /var/lib/asterisk/sounds/digits ; \
+	done
+	for x in sounds/vm-* sounds/transfer* ; do \
+		install $$x /var/lib/asterisk/sounds ; \
+	done
+
 install: all
 	mkdir -p $(MODULES_DIR)
 	for x in $(SUBDIRS); do $(MAKE) -C $$x install || exit 1 ; done
 	install -d /usr/include/asterisk
 	install include/asterisk/*.h /usr/include/asterisk
+	rm -f /var/lib/asterisk/sounds/vm
+	mkdir -p /var/spool/asterisk/vm
+	rm -f /usr/lib/asterisk/modules/chan_ixj.so
+	mkdir -p /var/lib/asterisk/sounds
+	( cd /var/lib/asterisk/sounds  ; ln -s ../../../spool/asterisk/vm . )
+	@echo " +---- Asterisk Installation Complete -------+"  
+	@echo " + Asterisk has successfully been installed. +"  
+	@echo " + If you would like to install the sample   +"  
+	@echo " + configuration files (overwriting any      +"
+	@echo " + existing config files), run:              +"  
+	@echo " +                                           +"
+	@echo " +               make samples                +"
+	@echo " +                                           +"
+	@echo " +-------------------------------------------+"  
+
+samples: all datafiles
+	mkdir -p /etc/asterisk
+	for x in configs/*.sample; do \
+		if [ -f /etc/asterisk/`basename $$x .sample` ]; then \
+			mv -f /etc/asterisk/`basename $$x .sample` /etc/asterisk/`basename $$x .sample`.old ; \
+		fi ; \
+		install $$x /etc/asterisk/`basename $$x .sample` ;\
+	done
+	for x in sounds/demo-*; do \
+		install $$x /var/lib/asterisk/sounds; \
+	done
+	mkdir -p /var/spool/asterisk/vm/1234/INBOX
+	:> /var/lib/asterisk/sounds/vm/1234/unavail.gsm
+	for x in vm-theperson digits/1 digits/2 digits/3 digits/4 vm-isunavail; do \
+		cat /var/lib/asterisk/sounds/$$x.gsm >> /var/lib/asterisk/sounds/vm/1234/unavail.gsm ; \
+	done
+	:> /var/lib/asterisk/sounds/vm/1234/busy.gsm
+	for x in vm-theperson digits/1 digits/2 digits/3 digits/4 vm-isonphone; do \
+		cat /var/lib/asterisk/sounds/$$x.gsm >> /var/lib/asterisk/sounds/vm/1234/busy.gsm ; \
+	done

README

@@ -1,6 +1,6 @@
 The Asterisk Open Source PBX
 by Mark Spencer <markster@linux-support.net>
-Copyright (C) 1999, Linux Support Services, LLC and Adtran, Inc.
+Copyright (C) 1999, Mark Spencer
 ================================================================
 * WHAT IS ASTERISK
   Asterisk is an Open Source PBX and telephony toolkit.  It is, in a
@@ -22,6 +22,60 @@ as well.
 
   If you want to use format_wav module, then you need a very recent
 version of libaudiofile (at least version 0.2.0, or you can apply the
-following patch to version 0.1.9):
+included patch.  RPMS for the patched libaudiofile are available at:
+ftp://ftp.asteriskpbx.com/pub/asterisk/support
 
 
+* GETTING STARTED
+
+First, be sure you've installed the required libaudiofile upgrade if 
+you want to use the non-GSM WAV format.  Next, be sure you've got
+supported hardware.  To use Asterisk right now, you will need one of 
+the following:
+
+	* Adtran Atlas 800 Plus
+	* QuickNet Internet PhoneJack
+	* Full Duplex Sound Card supported by Linux
+
+Assuming you have one of these (most likely the third) you're ready to 
+proceed:
+
+1) Run "make"
+2) Run "make install"
+
+If this is your first time working with Asterisk, you may wish to install
+the sample PBX, with demonstration extensions, etc.  If so, run:
+
+	"make samples"
+
+Doing so will overwrite any existing config files you have.
+
+Finally, you can launch Asterisk with:
+
+	./asterisk -vvvc
+
+If you get an error about unresolved symbols, install the updated
+libaudiofile (available at ftp://ftp.asteriskpbx.com/pub/asterisk/support
+
+You'll see a bunch of verbose messages fly by your screen as Asterisk
+initializes (that's the "very very verbose" mode).  When it's ready, if
+you specified the "c" then you'll get a command line console, that looks
+like this: 
+
+*CLI>
+
+You can type "help" at any time to get help with the system.  For help
+with a specific command, type "help <command>".  To start the PBX using
+your sound card, you can type "dial" to dial the PBX.  Then you can use
+"answer", "hangup", and "dial" to simulate the actions of a telephone.
+Remember that if you don't have a full duplex sound card (And asterisk
+will tell you somewhere in its verbose messages if you do/don't) than it
+won't work right (not yet).
+
+Feel free to look over the configuration files in /etc/asterisk, where
+you'll find a lot of information about what you can do with Asterisk.
+
+Finally, you may wish to visit the web site and join the mailing list if
+you're interested in getting more information.
+
+Mark

apps/Makefile

@@ -3,7 +3,7 @@
 # 
 # Makefile for PBX frontends (dynamically loaded)
 #
-# Copyright (C) 1999, Adtran Inc. and Linux Support Services, LLC
+# Copyright (C) 1999, Mark Spencer
 #
 # Mark Spencer <markster@linux-support.net>
 #
@@ -11,7 +11,7 @@
 # the GNU General Public License
 #
 
-APPS=app_dial.so app_playback.so app_voicemail.so app_directory.so app_intercom.so app_mp3.so
+APPS=app_dial.so app_playback.so app_voicemail.so app_directory.so app_intercom.so app_mp3.so app_system.so app_echo.so
 
 CFLAGS+=
 

apps/app_dial.c

@@ -3,7 +3,7 @@
  *
  * Trivial application to dial a channel
  * 
- * Copyright (C) 1999, Adtran Inc. and Linux Support Services, LLC
+ * Copyright (C) 1999, Mark Spencer
  *
  * Mark Spencer <markster@linux-support.net>
  *
@@ -17,6 +17,8 @@
 #include <asterisk/pbx.h>
 #include <asterisk/options.h>
 #include <asterisk/module.h>
+#include <asterisk/translate.h>
+#include <asterisk/say.h>
 #include <stdlib.h>
 #include <errno.h>
 #include <unistd.h>
@@ -24,14 +26,31 @@
 #include <stdlib.h>
 #include <stdio.h>
 #include <sys/time.h>
+#include <sys/signal.h>
 
 #include <pthread.h>
 
-
-static char *tdesc = "Trivial Dialing Application";
+static char *tdesc = "Dialing/Parking Application";
 
 static char *app = "Dial";
 
+static char *parkedcall = "ParkedCall";
+
+/* No more than 90 seconds parked before you do something with them */
+static int parkingtime = 90000;
+
+/* Context for which parking is made accessible */
+static char parking_con[AST_MAX_EXTENSION] = "default";
+
+/* Extension you type to park the call */
+static char parking_ext[AST_MAX_EXTENSION] = "700";
+
+/* First available extension for parking */
+static int parking_start = 701;
+
+/* Last available extension for parking */
+static int parking_stop = 750;
+
 /* We define a customer "local user" structure because we
    use it not only for keeping track of what is in use but
    also for keeping track of who we're dialing. */
@@ -43,8 +62,164 @@ struct localuser {
 	struct localuser *next;
 };
 
+struct parkeduser {
+	struct ast_channel *chan;
+	struct timeval start;
+	int parkingnum;
+	/* Where to go if our parking time expires */
+	char context[AST_MAX_EXTENSION];
+	char exten[AST_MAX_EXTENSION];
+	int priority;
+	struct parkeduser *next;
+};
+
+static struct parkeduser *parkinglot;
+
+static pthread_mutex_t parking_lock = PTHREAD_MUTEX_INITIALIZER;
+
+static pthread_t parking_thread;
+
 LOCAL_USER_DECL;
 
+static int bridge_call(struct ast_channel *chan, struct ast_channel *peer, int allowredirect);
+
+
+static int park_call(struct ast_channel *chan, struct ast_channel *peer)
+{
+	/* We put the user in the parking list, then wake up the parking thread to be sure it looks
+	   after these channels too */
+	struct parkeduser *pu, *cur;
+	int x;
+	pu = malloc(sizeof(struct parkeduser));
+	if (pu) {
+		pthread_mutex_lock(&parking_lock);
+		for (x=parking_start;x<=parking_stop;x++) {
+			cur = parkinglot;
+			while(cur) {
+				if (cur->parkingnum == x) 
+					break;
+				cur = cur->next;
+			}
+			if (!cur)
+				break;
+		}
+		if (x <= parking_stop) {
+			pu->chan = chan;
+			gettimeofday(&pu->start, NULL);
+			pu->parkingnum = x;
+			/* Remember what had been dialed, so that if the parking
+			   expires, we try to come back to the same place */
+			strncpy(pu->context, chan->context, sizeof(pu->context));
+			strncpy(pu->exten, chan->exten, sizeof(pu->exten));
+			pu->priority = chan->priority;
+			pu->next = parkinglot;
+			parkinglot = pu;
+			pthread_mutex_unlock(&parking_lock);
+			/* Wake up the (presumably select()ing) thread */
+			pthread_kill(parking_thread, SIGURG);
+			if (option_verbose > 1) 
+				ast_verbose(VERBOSE_PREFIX_2 "Parked %s on %d\n", pu->chan->name, pu->parkingnum);
+			ast_say_digits(peer, pu->parkingnum, peer->language);
+			return 0;
+		} else {
+			ast_log(LOG_WARNING, "No more parking spaces\n");
+			free(pu);
+			pthread_mutex_unlock(&parking_lock);
+			return -1;
+		}
+	} else {
+		ast_log(LOG_WARNING, "Out of memory\n");
+		return -1;
+	}
+	return 0;
+}
+
+static void *do_parking_thread(void *ignore)
+{
+	int ms, tms, max;
+	struct parkeduser *pu, *pl, *pt = NULL;
+	struct timeval tv;
+	struct ast_frame *f;
+	fd_set rfds, efds;
+	fd_set nrfds, nefds;
+	FD_ZERO(&rfds);
+	FD_ZERO(&efds);
+	for (;;) {
+		ms = -1;
+		max = -1;
+		pthread_mutex_lock(&parking_lock);
+		pl = NULL;
+		pu = parkinglot;
+		gettimeofday(&tv, NULL);
+		FD_ZERO(&nrfds);
+		FD_ZERO(&nefds);
+		while(pu) {
+			tms = (tv.tv_sec - pu->start.tv_sec) * 1000 + (tv.tv_usec - pu->start.tv_usec) / 1000;
+			if (tms > parkingtime) {
+				/* They've been waiting too long, send them back to where they came.  Theoretically they
+				   should have their original extensions and such, but we copy to be on the safe side */
+				strncpy(pu->chan->exten, pu->exten, sizeof(pu->chan->exten));
+				strncpy(pu->chan->context, pu->context, sizeof(pu->chan->context));
+				pu->chan->priority = pu->priority;
+				/* Start up the PBX, or hang them up */
+				if (ast_pbx_start(pu->chan))  {
+					ast_log(LOG_WARNING, "Unable to restart the PBX for user on '%s', hanging them up...\n", pu->chan->name);
+					ast_hangup(pu->chan);
+				}
+				/* And take them out of the parking lot */
+				if (pl) 
+					pl->next = pu->next;
+				else
+					parkinglot = pu->next;
+				pt = pu;
+				pu = pu->next;
+				free(pt);
+			} else if (FD_ISSET(pu->chan->fd, &rfds) || FD_ISSET(pu->chan->fd, &efds)) {
+				/* See if they need servicing */
+				f = ast_read(pu->chan);
+				if (!f || ((f->frametype == AST_FRAME_CONTROL) && (f->subclass ==  AST_CONTROL_HANGUP))) {
+					/* There's a problem, hang them up*/
+					if (option_verbose > 1) 
+						ast_verbose(VERBOSE_PREFIX_2 "%s got tired of being parked\n", pu->chan->name);
+					ast_hangup(pu->chan);
+					/* And take them out of the parking lot */
+					if (pl) 
+						pl->next = pu->next;
+					else
+						parkinglot = pu->next;
+					pt = pu;
+					pu = pu->next;
+					free(pt);
+				} else {
+					/* XXX Maybe we could do something with packets, like dial "0" for operator or something XXX */
+					ast_frfree(f);
+					goto std;	/* XXX Ick: jumping into an else statement??? XXX */
+				}
+			} else  {
+				/* Keep this one for next one */
+std:			FD_SET(pu->chan->fd, &nrfds);
+				FD_SET(pu->chan->fd, &nefds);
+				/* Keep track of our longest wait */
+				if ((tms < ms) || (ms < 0))
+					ms = tms;
+				if (pu->chan->fd > max)
+					max = pu->chan->fd;
+				pl = pu;
+				pu = pu->next;
+			}
+		}
+		pthread_mutex_unlock(&parking_lock);
+		rfds = nrfds;
+		efds = nefds;
+		tv.tv_sec = ms / 1000;
+		tv.tv_usec = (ms % 1000) * 1000;
+		/* Wait for something to happen */
+		select(max + 1, &rfds, NULL, &efds, (ms > -1) ? &tv : NULL);
+		pthread_testcancel();
+	}
+	return NULL;	/* Never reached */
+}
+
 static void hanguptree(struct localuser *outgoing, struct ast_channel *exception)
 {
 	/* Hang up a tree of stuff */
@@ -205,6 +380,8 @@ static int bridge_call(struct ast_channel *chan, struct ast_channel *peer, int a
 	if (chan->state != AST_STATE_UP)
 		if (ast_answer(chan))
 			return -1;
+	peer->appl = "Bridged Call";
+	peer->data = chan->name;
 	cs[0] = chan;
 	cs[1] = peer;
 	for (/* ever */;;) {
@@ -214,18 +391,24 @@ static int bridge_call(struct ast_channel *chan, struct ast_channel *peer, int a
 			continue;
 		}
 		f = ast_read(who);
-		if (!f || ((f->frametype == AST_FRAME_CONTROL) && (f->subclass == AST_CONTROL_HANGUP))) 
+		if (!f || ((f->frametype == AST_FRAME_CONTROL) && 
+					((f->subclass == AST_CONTROL_HANGUP) ||
+					 (f->subclass == AST_CONTROL_BUSY)))) 
 			return -1;
 		if ((f->frametype == AST_FRAME_VOICE) ||
-		    (f->frametype == AST_FRAME_DTMF)) {
-			if ((f->frametype == AST_FRAME_DTMF) && (who == peer) && allowredirect) {
+		    (f->frametype == AST_FRAME_DTMF) ||
+			(f->frametype == AST_FRAME_TEXT)) {
+			if ((f->frametype == AST_FRAME_DTMF) && (who == peer) && allowredirect &&
+			     (f->subclass == '#')) {
 				if (f->subclass == '#') {
 					memset(newext, 0, sizeof(newext));
 					ptr = newext;
 					len = ast_pbx_longest_extension(chan->context) + 1;
+					if (len < ast_pbx_longest_extension("default") + 1)
+						len = ast_pbx_longest_extension("default") + 1;
 
 					/* Transfer */
-					if ((res=ast_streamfile(peer, "pbx-transfer")))
+					if ((res=ast_streamfile(peer, "pbx-transfer", chan->language)))
 						break;
 					if ((res=ast_waitstream(peer, AST_DIGIT_ANY)) < 0)
 						break;
@@ -239,15 +422,35 @@ static int bridge_call(struct ast_channel *chan, struct ast_channel *peer, int a
 					res = ast_readstring(peer, ptr, len, 3000, 2000, "#");
 					if (res)
 						break;
-					if (ast_exists_extension(chan, chan->context, newext, 1)) {
-						/* Set the channel's new extension, since it exists */
+					if (!strcmp(newext, parking_ext)) {
+						if (!park_call(chan, peer)) {
+							/* We return non-zero, but tell the PBX not to hang the channel when
+							   the thread dies -- We have to be careful now though.  We are responsible for 
+							   hanging up the channel, else it will never be hung up! */
+							res=AST_PBX_KEEPALIVE;
+							break;
+						} else {
+							ast_log(LOG_WARNING, "Unable to park call %s\n", chan->name);
+						}
+						/* XXX Maybe we should have another message here instead of invalid extension XXX */
+					} else if (ast_exists_extension(chan, peer->context, newext, 1)) {
+						/* Set the channel's new extension, since it exists, using peer context */
 						strncpy(chan->exten, newext, sizeof(chan->exten));
+						strncpy(chan->context, peer->context, sizeof(chan->context));
+						chan->priority = 0;
+						ast_frfree(f);
+						res=0;
+						break;
+					} else if (ast_exists_extension(chan, "default", newext, 1)) {
+						/* Set the channel's new extension, since it exists, using peer context */
+						strncpy(chan->exten, newext, sizeof(chan->exten));
+						strncpy(chan->context, "default", sizeof(chan->context));
 						chan->priority = 0;
 						ast_frfree(f);
 						res=0;
 						break;
 					}
-					res = ast_streamfile(peer, "pbx-invalid");
+					res = ast_streamfile(peer, "pbx-invalid", chan->language);
 					if (res)
 						break;
 					res = ast_waitstream(peer, AST_DIGIT_ANY);
@@ -275,15 +478,75 @@ static int bridge_call(struct ast_channel *chan, struct ast_channel *peer, int a
 	return res;
 }
 
+static int park_exec(struct ast_channel *chan, void *data)
+{
+	int res=0;
+	struct localuser *u;
+	struct ast_channel *peer=NULL, *nchan;
+	struct parkeduser *pu, *pl=NULL;
+	int park;
+	if (!data) {
+		ast_log(LOG_WARNING, "Park requires an argument (extension number)\n");
+		return -1;
+	}
+	LOCAL_USER_ADD(u);
+	park = atoi((char *)data);
+	pthread_mutex_lock(&parking_lock);
+	pu = parkinglot;
+	while(pu) {
+		if (pu->parkingnum == park) {
+			if (pl)
+				pl->next = pu->next;
+			else
+				parkinglot = pu->next;
+			break;
+		}
+		pu = pu->next;
+	}
+	pthread_mutex_unlock(&parking_lock);
+	if (pu) {
+		peer = pu->chan;
+		free(pu);
+	}
+	if (peer) {
+		/* Build a translator if necessary */
+		if (peer->format & chan->format) 
+			nchan = chan;
+		else
+			nchan = ast_translator_create(chan, peer->format, AST_DIRECTION_BOTH);
+		/* This runs sorta backwards, since we give the incoming channel control, as if it
+		   were the person called. */
+		if (option_verbose > 2) 
+			ast_verbose(VERBOSE_PREFIX_3 "Channel %s connected to parked call %d\n", chan->name, park);
+		res = bridge_call(peer, nchan, 1);
+		if (nchan != chan)
+			ast_translator_destroy(nchan);
+		/* Simulate the PBX hanging up */
+		if (res != AST_PBX_KEEPALIVE)
+			ast_hangup(peer);
+		return -1;
+	} else {
+		/* XXX Play a message XXX */
+		if (option_verbose > 2) 
+			ast_verbose(VERBOSE_PREFIX_3 "Channel %s tried to talk to non-existant parked call %d\n", chan->name, park);
+		res = -1;
+	}
+	LOCAL_USER_REMOVE(u);
+	return res;
+}
+
 static int dial_exec(struct ast_channel *chan, void *data)
 {
 	int res=-1;
 	struct localuser *u;
 	char *info, *peers, *timeout, *tech, *number, *rest, *cur;
 	struct localuser *outgoing=NULL, *tmp;
-	struct ast_channel *peer;
+	struct ast_channel *peer, *npeer;
 	int to;
 	int allowredir=0;
+	char numsubst[AST_MAX_EXTENSION];
+	char restofit[AST_MAX_EXTENSION];
+	char *newnum;
 	
 	if (!data) {
 		ast_log(LOG_WARNING, "Dial requires an argument (technology1/number1&technology2/number2...|optional timeout)\n");
@@ -292,8 +555,9 @@ static int dial_exec(struct ast_channel *chan, void *data)
 	
 	LOCAL_USER_ADD(u);
 	
-	/* Parse our arguments */
-	info = strdup((char *)data);
+	/* Parse our arguments XXX Check for failure XXX */
+	info = malloc(strlen((char *)data) + AST_MAX_EXTENSION);
+	strncpy(info, (char *)data, strlen((char *)data) + AST_MAX_EXTENSION);
 	peers = strtok(info, "|");
 	if (!peers) {
 		ast_log(LOG_WARNING, "Dial argument takes format (technology1/number1&technology2/number2...|optional timeout)\n");
@@ -305,11 +569,11 @@ static int dial_exec(struct ast_channel *chan, void *data)
 		cur = strtok(rest, "&");
 		/* Remember where to start next time */
 		rest = strtok(NULL, "\128");
-		/* Get a technology/number pair */
+		/* Get a technology/[device:]number pair */
 		tech = strtok(cur, "/");
 		number = strtok(NULL, "&");
 		if (!number) {
-			ast_log(LOG_WARNING, "Dial argument takes format (technology1/number1&technology2/number2...|optional timeout)\n");
+			ast_log(LOG_WARNING, "Dial argument takes format (technology1/[device:]number1&technology2/[device:]number2...|optional timeout)\n");
 			goto out;
 		}
 		tmp = malloc(sizeof(struct localuser));
@@ -318,36 +582,40 @@ static int dial_exec(struct ast_channel *chan, void *data)
 			goto out;
 		}
 		tmp->allowredirect = 1;
+		strncpy(numsubst, number, sizeof(numsubst));
 		/* If we're dialing by extension, look at the extension to know what to dial */
-		if (!strcasecmp(number, "BYEXTENSION")) {
-			if (option_debug)
-				ast_log(LOG_DEBUG, "Dialing by extension %s\n", chan->exten);
-			number = chan->exten;
+		if ((newnum = strstr(numsubst, "BYEXTENSION"))) {
+			strncpy(restofit, newnum + strlen("BYEXTENSION"), sizeof(restofit));
+			snprintf(newnum, sizeof(numsubst) - (newnum - numsubst), "%s%s", chan->exten,restofit);
 			/* By default, if we're dialing by extension, don't permit redirecting */
 			tmp->allowredirect = 0;
+			if (option_debug)
+				ast_log(LOG_DEBUG, "Dialing by extension %s\n", numsubst);
 		}
 		/* Request the peer */
-		tmp->chan = ast_request(tech, chan->format, number);
+		tmp->chan = ast_request(tech, chan->format, numsubst);
 		if (!tmp->chan) {
 			/* If we can't, just go on to the next call */
 			ast_log(LOG_WARNING, "Unable to create channel of type '%s'\n", tech);
 			free(tmp);
 			continue;
 		}
+		tmp->chan->appl = "AppDial";
+		tmp->chan->data = "(Outgoing Line)";
 		/* Place the call, but don't wait on the answer */
-		res = ast_call(tmp->chan, number, 0);
+		res = ast_call(tmp->chan, numsubst, 0);
 		if (res) {
 			/* Again, keep going even if there's an error */
 			if (option_debug)
 				ast_log(LOG_DEBUG, "ast call on peer returned %d\n", res);
 			else if (option_verbose > 2)
-				ast_verbose(VERBOSE_PREFIX_3 "Couldn't call %s\n", number);
+				ast_verbose(VERBOSE_PREFIX_3 "Couldn't call %s\n", numsubst);
 			ast_hangup(tmp->chan);
 			free(tmp);
 			continue;
 		} else
 			if (option_verbose > 2)
-				ast_verbose(VERBOSE_PREFIX_3 "Called %s\n", number);
+				ast_verbose(VERBOSE_PREFIX_3 "Called %s\n", numsubst);
 		/* Put them in the list of outgoing thingies...  We're ready now. 
 		   XXX If we're forcibly removed, these outgoing calls won't get
 		   hung up XXX */
@@ -376,7 +644,14 @@ static int dial_exec(struct ast_channel *chan, void *data)
 		   conversation.  */
 		hanguptree(outgoing, peer);
 		outgoing = NULL;
-		res = bridge_call(chan, peer, allowredir);
+		/* Build a translator if necessary */
+		if (peer->format & chan->format) 
+			npeer = peer;
+		else
+			npeer = ast_translator_create(peer, chan->format, AST_DIRECTION_BOTH);
+		res = bridge_call(chan, npeer, allowredir);
+		if (npeer != peer)
+			ast_translator_destroy(npeer);
 		ast_hangup(peer);
 	}	
 out:
@@ -394,7 +669,24 @@ int unload_module(void)
 
 int load_module(void)
 {
-	return ast_register_application(app, dial_exec);
+	int res;
+	int x;
+	struct ast_context *con;
+	char exten[AST_MAX_EXTENSION];
+	con = ast_context_find(parking_con);
+	if (!con) {
+		ast_log(LOG_ERROR, "Parking context '%s' does not exist\n", parking_con);
+		return -1;
+	}
+	for(x=parking_start; x<=parking_stop;x++) {
+		snprintf(exten, sizeof(exten), "%d", x);
+		ast_add_extension2(con, 1, exten, 1, parkedcall, strdup(exten), free);
+	}
+	pthread_create(&parking_thread, NULL, do_parking_thread, NULL);
+	res = ast_register_application(parkedcall, park_exec);
+	if (!res)
+		res = ast_register_application(app, dial_exec);
+	return res;
 }
 
 char *description(void)

apps/app_directory.c

@@ -3,7 +3,7 @@
  *
  * Provide a directory of extensions
  * 
- * Copyright (C) 1999, Adtran Inc. and Linux Support Services, LLC
+ * Copyright (C) 1999, Mark Spencer
  *
  * Mark Spencer <markster@linux-support.net>
  *
@@ -156,17 +156,17 @@ static int do_directory(struct ast_channel *chan, struct ast_config *cfg, char *
 			if (v) {
 				/* We have a match -- play a greeting if they have it */
 				snprintf(fn, sizeof(fn), "%s/vm/%s/greet", AST_SPOOL_DIR, v->name);
-				if (ast_fileexists(fn, NULL)) {
-					res = ast_streamfile(chan, fn);
+				if (ast_fileexists(fn, NULL, chan->language)) {
+					res = ast_streamfile(chan, fn, chan->language);
 					if (!res)
 						res = ast_waitstream(chan, AST_DIGIT_ANY);
 					ast_stopstream(chan);
 				} else {
-					res = ast_say_digit_str(chan, v->name);
+					res = ast_say_digit_str(chan, v->name, chan->language);
 				}
 ahem:
 				if (!res)
-					res = ast_streamfile(chan, "dir-instr");
+					res = ast_streamfile(chan, "dir-instr", chan->language);
 				if (!res)
 					res = ast_waitstream(chan, AST_DIGIT_ANY);
 				if (!res)
@@ -189,9 +189,9 @@ ahem:
 				}
 			} else {
 				if (found) 
-					res = ast_streamfile(chan, "dir-nomore");
+					res = ast_streamfile(chan, "dir-nomore", chan->language);
 				else
-					res = ast_streamfile(chan, "dir-nomatch");
+					res = ast_streamfile(chan, "dir-nomatch", chan->language);
 				if (!res)
 					res = 1;
 				return res;
@@ -219,7 +219,7 @@ static int directory_exec(struct ast_channel *chan, void *data)
 	LOCAL_USER_ADD(u);
 top:
 	if (!res)
-		res = ast_streamfile(chan, "dir-intro");
+		res = ast_streamfile(chan, "dir-intro", chan->language);
 	if (!res)
 		res = ast_waitstream(chan, AST_DIGIT_ANY);
 	ast_stopstream(chan);

apps/app_echo.c

@@ -0,0 +1,80 @@
+/*
+ * Asterisk -- A telephony toolkit for Linux.
+ *
+ * Echo application -- play back what you hear to evaluate latency
+ * 
+ * Copyright (C) 1999, Mark Spencer
+ *
+ * Mark Spencer <markster@linux-support.net>
+ *
+ * This program is free software, distributed under the terms of
+ * the GNU General Public License
+ */
+
+#include <asterisk/file.h>
+#include <asterisk/logger.h>
+#include <asterisk/channel.h>
+#include <asterisk/pbx.h>
+#include <asterisk/module.h>
+#include <stdlib.h>
+#include <unistd.h>
+#include <string.h>
+#include <stdlib.h>
+
+#include <pthread.h>
+
+
+static char *tdesc = "Simple Echo Application";
+
+static char *app = "Echo";
+
+STANDARD_LOCAL_USER;
+
+LOCAL_USER_DECL;
+
+static int skel_exec(struct ast_channel *chan, void *data)
+{
+	int res=-1;
+	struct localuser *u;
+	struct ast_frame *f;
+	LOCAL_USER_ADD(u);
+	/* Do our thing here */
+	while((f = ast_read(chan))) {
+		if (f->frametype == AST_FRAME_VOICE) {
+			if (ast_write(chan, f)) 
+				break;
+		} else if (f->frametype == AST_FRAME_DTMF) {
+			if (f->subclass == '#') {
+				res = 0;
+				break;
+			} else
+				if (ast_write(chan, f))
+					break;
+		}
+	}
+	LOCAL_USER_REMOVE(u);
+	return res;
+}
+
+int unload_module(void)
+{
+	STANDARD_HANGUP_LOCALUSERS;
+	return ast_unregister_application(app);
+}
+
+int load_module(void)
+{
+	return ast_register_application(app, skel_exec);
+}
+
+char *description(void)
+{
+	return tdesc;
+}
+
+int usecount(void)
+{
+	int res;
+	STANDARD_USECOUNT(res);
+	return res;
+}

apps/app_intercom.c

@@ -3,7 +3,7 @@
  *
  * Use /dev/dsp as an intercom.
  * 
- * Copyright (C) 1999, Adtran Inc. and Linux Support Services, LLC
+ * Copyright (C) 1999, Mark Spencer
  *
  * Mark Spencer <markster@linux-support.net>
  *
@@ -59,7 +59,7 @@ static int write_audio(short *data, int len)
 		pthread_mutex_unlock(&sound_lock);
         return -1;
     }
-		res = write(sound, data, len);
+	res = write(sound, data, len);
 	pthread_mutex_unlock(&sound_lock);
 	return res;
 }
@@ -97,7 +97,7 @@ static int create_audio()
 		return -1;
 	}
 	if (fmt != desired) {
-		ast_log(LOG_WARNING, "Requested %d Hz, got %d Hz -- sound may be choppy\n");
+		ast_log(LOG_WARNING, "Requested %d Hz, got %d Hz -- sound may be choppy\n", desired, fmt);
 	}
 #if 1
 	/* 2 bytes * 15 units of 2^5 = 32 bytes per buffer */
@@ -146,7 +146,7 @@ static int intercom_exec(struct ast_channel *chan, void *data)
 									res = 0;
 							} else
 								ast_log(LOG_DEBUG, "Unable to handle non-signed linear frame (%d)\n", f->subclass);
-						}
+						} 
 					}
 					ast_frfree(f);
 				} else

apps/app_mp3.c

@@ -3,7 +3,7 @@
  *
  * Silly application to play an MP3 file -- uses mpg123
  * 
- * Copyright (C) 1999, Adtran Inc. and Linux Support Services, LLC
+ * Copyright (C) 1999, Mark Spencer
  *
  * Mark Spencer <markster@linux-support.net>
  *

apps/app_playback.c

@@ -3,7 +3,7 @@
  *
  * Trivial application to playback a sound file
  * 
- * Copyright (C) 1999, Adtran Inc. and Linux Support Services, LLC
+ * Copyright (C) 1999, Mark Spencer
  *
  * Mark Spencer <markster@linux-support.net>
  *
@@ -31,18 +31,24 @@ LOCAL_USER_DECL;
 
 static int playback_exec(struct ast_channel *chan, void *data)
 {
-	int res;
+	int res = 0;
 	struct localuser *u;
 	if (!data) {
 		ast_log(LOG_WARNING, "Playback requires an argument (filename)\n");
 		return -1;
 	}
 	LOCAL_USER_ADD(u);
-	ast_stopstream(chan);
-	res = ast_streamfile(chan, (char *)data);
-	if (!res)
-		res = ast_waitstream(chan, "");
-	ast_stopstream(chan);
+	if (chan->state != AST_STATE_UP)
+		res = ast_answer(chan);
+	if (!res) {
+		ast_stopstream(chan);
+		res = ast_streamfile(chan, (char *)data, chan->language);
+		if (!res) 
+			res = ast_waitstream(chan, "");
+		else
+			ast_log(LOG_WARNING, "ast_streamfile failed on %s for %s\n", chan->name, (char *)data);
+		ast_stopstream(chan);
+	}
 	LOCAL_USER_REMOVE(u);
 	return res;
 }

apps/app_skel.c

@@ -3,7 +3,7 @@
  *
  * Skeleton application
  * 
- * Copyright (C) 1999, Adtran Inc. and Linux Support Services, LLC
+ * Copyright (C) 1999, Mark Spencer
  *
  * Mark Spencer <markster@linux-support.net>
  *
@@ -15,6 +15,7 @@
 #include <asterisk/logger.h>
 #include <asterisk/channel.h>
 #include <asterisk/pbx.h>
+#include <asterisk/module.h>
 #include <stdlib.h>
 #include <unistd.h>
 #include <string.h>
@@ -22,63 +23,32 @@
 
 #include <pthread.h>
 
-static pthread_mutex_t skellock = PTHREAD_MUTEX_INITIALIZER;
-
-static int usecnt=0;
 
 static char *tdesc = "Trivial skeleton Application";
 
 static char *app = "skel";
 
-struct skeluser {
-	struct ast_channel *chan;
-	struct skeluser *next;
-} *users = NULL;
+STANDARD_LOCAL_USER;
+
+LOCAL_USER_DECL;
 
 static int skel_exec(struct ast_channel *chan, void *data)
 {
 	int res=0;
-	struct skeluser *u, *ul=NULL;
+	struct localuser *u;
 	if (!data) {
 		ast_log(LOG_WARNING, "skel requires an argument (filename)\n");
 		return -1;
 	}
-	if (!(u=malloc(sizeof(struct skeluser)))) {
-		ast_log(LOG_WARNING, "Out of memory\n");
-		return -1;
-	}
-	pthread_mutex_lock(&skellock);
-	u->chan = chan;
-	u->next = users;
-	users = u;
-	usecnt++;
-	pthread_mutex_unlock(&skellock);
+	LOCAL_USER_ADD(u);
 	/* Do our thing here */
-	pthread_mutex_lock(&skellock);
-	u = users;
-	while(u) {
-		if (ul)
-			ul->next = u->next;
-		else
-			users = u->next;
-		u = u->next;
-	}
-	usecnt--;
-	pthread_mutex_unlock(&skellock);
+	LOCAL_USER_REMOVE(u);
 	return res;
 }
 
 int unload_module(void)
 {
-	struct skeluser *u;
-	pthread_mutex_lock(&skellock);
-	u = users;
-	while(u) {
-		/* Hang up anybody who is using us */
-		ast_softhangup(u->chan);
-		u = u->next;
-	}
-	pthread_mutex_unlock(&skellock);
+	STANDARD_HANGUP_LOCALUSERS;
 	return ast_unregister_application(app);
 }
 
@@ -95,8 +65,6 @@ char *description(void)
 int usecount(void)
 {
 	int res;
-	pthread_mutex_lock(&skellock);
-	res = usecnt;
-	pthread_mutex_unlock(&skellock);
+	STANDARD_USECOUNT(res);
 	return res;
 }

apps/app_system.c

@@ -0,0 +1,82 @@
+/*
+ * Asterisk -- A telephony toolkit for Linux.
+ *
+ * Execute arbitrary system commands
+ * 
+ * Copyright (C) 1999, Mark Spencer
+ *
+ * Mark Spencer <markster@linux-support.net>
+ *
+ * This program is free software, distributed under the terms of
+ * the GNU General Public License
+ */
+
+#include <asterisk/file.h>
+#include <asterisk/logger.h>
+#include <asterisk/channel.h>
+#include <asterisk/pbx.h>
+#include <asterisk/module.h>
+#include <stdlib.h>
+#include <unistd.h>
+#include <string.h>
+#include <stdlib.h>
+
+#include <pthread.h>
+
+
+static char *tdesc = "Generic System() application";
+
+static char *app = "System";
+
+STANDARD_LOCAL_USER;
+
+LOCAL_USER_DECL;
+
+static int skel_exec(struct ast_channel *chan, void *data)
+{
+	int res=0;
+	struct localuser *u;
+	if (!data) {
+		ast_log(LOG_WARNING, "System requires an argument(command)\n");
+		return -1;
+	}
+	LOCAL_USER_ADD(u);
+	/* Do our thing here */
+	res = system((char *)data);
+	if (res < 0) {
+		ast_log(LOG_WARNING, "Unable to execute '%s'\n", data);
+		res = -1;
+	} else if (res == 127) {
+		ast_log(LOG_WARNING, "Unable to execute '%s'\n", data);
+		res = -1;
+	} else {
+		if (res && ast_exists_extension(chan, chan->context, chan->exten, chan->priority + 101)) 
+			chan->priority+=100;
+		res = 0;
+	}
+	LOCAL_USER_REMOVE(u);
+	return res;
+}
+
+int unload_module(void)
+{
+	STANDARD_HANGUP_LOCALUSERS;
+	return ast_unregister_application(app);
+}
+
+int load_module(void)
+{
+	return ast_register_application(app, skel_exec);
+}
+
+char *description(void)
+{
+	return tdesc;
+}
+
+int usecount(void)
+{
+	int res;
+	STANDARD_USECOUNT(res);
+	return res;
+}

apps/app_voicemail.c

@@ -3,7 +3,7 @@
  *
  * Voicemail System (did you ever think it could be so easy?)
  * 
- * Copyright (C) 1999, Adtran Inc. and Linux Support Services, LLC
+ * Copyright (C) 1999, Mark Spencer
  *
  * Mark Spencer <markster@linux-support.net>
  *
@@ -81,15 +81,15 @@ static int announce_message(struct ast_channel *chan, char *dir, int msgcnt)
 {
 	char *fn;
 	int res;
-	res = ast_streamfile(chan, "vm-message");
+	res = ast_streamfile(chan, "vm-message", chan->language);
 	if (!res) {
 		res = ast_waitstream(chan, AST_DIGIT_ANY);
 		if (!res) {
-			res = ast_say_number(chan, msgcnt+1);
+			res = ast_say_number(chan, msgcnt+1, chan->language);
 			if (!res) {
 				fn = get_fn(dir, msgcnt);
 				if (fn) {
-					res = ast_streamfile(chan, fn);
+					res = ast_streamfile(chan, fn, chan->language);
 					free(fn);
 				}
 			}
@@ -141,7 +141,7 @@ static int leave_voicemail(struct ast_channel *chan, char *ext, int silent)
 	struct ast_config *cfg;
 	char *copy, *name, *passwd, *email, *dir, *fmt, *fmts, *fn=NULL;
 	char comment[256];
-	struct ast_filestream *writer, *others[MAX_OTHER_FORMATS];
+	struct ast_filestream *writer=NULL, *others[MAX_OTHER_FORMATS];
 	char *sfmt[MAX_OTHER_FORMATS];
 	int res = -1, fmtcnt=0, x;
 	int msgnum;
@@ -153,7 +153,7 @@ static int leave_voicemail(struct ast_channel *chan, char *ext, int silent)
 		ast_log(LOG_WARNING, "No such configuration file %s\n", VOICEMAIL_CONFIG);
 		return -1;
 	}
-	if ((copy = ast_variable_retrieve(cfg, chan->context, ext))) {
+	if ((copy = ast_variable_retrieve(cfg, NULL, ext))) {
 		/* Make sure they have an entry in the config */
 		copy = strdup(copy);
 		passwd = strtok(copy, ",");
@@ -167,7 +167,7 @@ static int leave_voicemail(struct ast_channel *chan, char *ext, int silent)
 		if (mkdir(dir, 0700) && (errno != EEXIST))
 			ast_log(LOG_WARNING, "mkdir '%s' failed: %s\n", dir, strerror(errno));
 		/* Stream an info message */
-		if (silent || !ast_streamfile(chan, INTRO)) {
+		if (silent || !ast_streamfile(chan, INTRO, chan->language)) {
 			/* Wait for the message to finish */
 			if (silent || !ast_waitstream(chan, "")) {
 				fmt = ast_variable_retrieve(cfg, "general", "format");
@@ -182,8 +182,12 @@ static int leave_voicemail(struct ast_channel *chan, char *ext, int silent)
 						snprintf(comment, sizeof(comment), "Voicemail from %s to %s (%s) on %s\n",
 											(chan->callerid ? chan->callerid : "Unknown"), 
 											name, ext, chan->name);
+						if (ast_fileexists(fn, NULL, chan->language) > 0) {
+							msgnum++;
+							continue;
+						}
 						writer = ast_writefile(fn, fmt, comment, O_EXCL, 1 /* check for other formats */, 0700);
-						if (!writer && (errno != EEXIST))
+						if (!writer)
 							break;
 						msgnum++;
 					} while(!writer && (msgnum < MAXMSG));
@@ -221,9 +225,14 @@ static int leave_voicemail(struct ast_channel *chan, char *ext, int silent)
 								if (f->frametype == AST_FRAME_VOICE) {
 									/* Write the primary format */
 									res = ast_writestream(writer, f);
+									if (res) {
+										ast_log(LOG_WARNING, "Error writing primary frame\n");
+										break;
+									}
 									/* And each of the others */
-									for (x=0;x<fmtcnt;x++)
+									for (x=0;x<fmtcnt;x++) {
 										res |= ast_writestream(others[x], f);
+									}
 									ast_frfree(f);
 									/* Exit on any error */
 									if (res) {
@@ -259,7 +268,7 @@ static int leave_voicemail(struct ast_channel *chan, char *ext, int silent)
 						if (outmsg) {
 							if (outmsg > 1) {
 								/* Let them know it worked */
-								ast_streamfile(chan, "vm-msgsaved");
+								ast_streamfile(chan, "vm-msgsaved", chan->language);
 								ast_waitstream(chan, "");
 							}
 							/* Send e-mail if applicable */
@@ -314,17 +323,31 @@ static int vm_execmain(struct ast_channel *chan, void *data)
 		ast_log(LOG_WARNING, "No voicemail configuration\n");
 		goto out;
 	}
-	if (ast_streamfile(chan, "vm-login"))
+	if (ast_streamfile(chan, "vm-login", chan->language)) {
+		ast_log(LOG_WARNING, "Couldn't stream login file\n");
 		goto out;
+	}
 	do {
 		/* Prompt for, and read in the username */
-		if (ast_readstring(chan, username, sizeof(username), 2000, 5000, "#"))
+		if (ast_readstring(chan, username, sizeof(username), 2000, 10000, "#")) {
+			ast_log(LOG_WARNING, "Couldn't read username\n");
 			goto out;
-		if (ast_streamfile(chan, "vm-password"))
+		}			
+		if (!strlen(username)) {
+			if (option_verbose > 2)
+				ast_verbose(VERBOSE_PREFIX_3 "Username not entered\n");
+			res = 0;
 			goto out;
-		if (ast_readstring(chan, password, sizeof(password), 2000, 5000, "#"))
+		}
+		if (ast_streamfile(chan, "vm-password", chan->language)) {
+			ast_log(LOG_WARNING, "Unable to stream password file\n");
 			goto out;
-		copy = ast_variable_retrieve(cfg, chan->context, username);
+		}
+		if (ast_readstring(chan, password, sizeof(password), 2000, 10000, "#")) {
+			ast_log(LOG_WARNING, "Unable to read password\n");
+			goto out;
+		}
+		copy = ast_variable_retrieve(cfg, NULL, username);
 		if (copy) {
 			copy = strdup(copy);
 			strtok(copy, ",");
@@ -336,7 +359,7 @@ static int vm_execmain(struct ast_channel *chan, void *data)
 		} else if (option_verbose > 2)
 			ast_verbose( VERBOSE_PREFIX_3 "No such user '%s' in config file\n", username);
 		if (!valid) {
-			if (ast_streamfile(chan, "vm-incorrect"))
+			if (ast_streamfile(chan, "vm-incorrect", chan->language))
 				break;
 			if (ast_waitstream(chan, ""))
 				break;
@@ -352,13 +375,13 @@ static int vm_execmain(struct ast_channel *chan, void *data)
 		   not deleted. */
 		do {
 			fn = get_fn(dir, maxmsg);
-			if ((res = ast_fileexists(fn, NULL))>0) {
+			if ((res = ast_fileexists(fn, NULL, chan->language))>0) {
 				maxmsg++;
 				deleted[maxmsg] = 0;
 			}
 			free(fn);
 		} while(res > 0);
-		if (ast_streamfile(chan, "vm-youhave"))
+		if (ast_streamfile(chan, "vm-youhave", chan->language))
 			goto out;
 		if ((d=ast_waitstream(chan, AST_DIGIT_ANY)) < 0)
 			goto out;
@@ -366,10 +389,10 @@ static int vm_execmain(struct ast_channel *chan, void *data)
 		if (!d) {
 			/* If they haven't interrupted us, play the message count */
 			if (maxmsg > 0) {
-				if ((d = ast_say_number(chan, maxmsg)) < 0)
+				if ((d = ast_say_number(chan, maxmsg, chan->language)) < 0)
 					goto out;
 			} else {
-				if (ast_streamfile(chan, "vm-no"))
+				if (ast_streamfile(chan, "vm-no", chan->language))
 					goto out;
 				if ((d=ast_waitstream(chan, AST_DIGIT_ANY)) < 0)
 					goto out;
@@ -377,7 +400,7 @@ static int vm_execmain(struct ast_channel *chan, void *data)
 			}
 			if (!d) {
 				/* And if they still haven't, give them the last word */
-				if (ast_streamfile(chan, ((maxmsg == 1) ? "vm-message" : "vm-messages")))
+				if (ast_streamfile(chan, ((maxmsg == 1) ? "vm-message" : "vm-messages"), chan->language))
 					goto out;
 				if (ast_waitstream(chan, AST_DIGIT_ANY) < 0)
 					goto out;
@@ -393,7 +416,7 @@ static int vm_execmain(struct ast_channel *chan, void *data)
 		ast_log(LOG_EVENT, "User '%s' logged in on channel '%s' with %d message(s).\n", username, chan->name, maxmsg);
 		if (option_verbose > 2)
 			ast_verbose( VERBOSE_PREFIX_3 "User '%s' logged in on channel %s with %d messages\n", username, chan->name, maxmsg);
-		if (!ast_streamfile(chan, "vm-instructions")) {
+		if (!ast_streamfile(chan, "vm-instructions", chan->language)) {
 			for(;;) {
 				if (chan->stream || (chan->trans && chan->trans->stream)) {
 					d = ast_waitstream(chan, AST_DIGIT_ANY);
@@ -401,7 +424,7 @@ static int vm_execmain(struct ast_channel *chan, void *data)
 					if (!d && (state == STATE_MESSAGE_PLAYING)) {
 						state  = STATE_MESSAGE;
 						/* If it runs out playing a message, then give directions */
-						if (!(d = ast_streamfile(chan, "vm-msginstruct")))
+						if (!(d = ast_streamfile(chan, "vm-msginstruct", chan->language)))
 							d = ast_waitstream(chan, AST_DIGIT_ANY);
 						ast_stopstream(chan);
 					}
@@ -417,12 +440,12 @@ restart:
 					   best based up on where they are.  Ditto if they press the '*' key. */
 					switch(state) {
 					case STATE_STARTING:
-						if (ast_streamfile(chan, "vm-instructions"))
+						if (ast_streamfile(chan, "vm-instructions", chan->language))
 							goto out;
 						break;
 					case STATE_MESSAGE:
 					case STATE_MESSAGE_PLAYING:
-						if (ast_streamfile(chan, "vm-msginstruct"))
+						if (ast_streamfile(chan, "vm-msginstruct", chan->language))
 							goto out;
 						break;
 					default:
@@ -437,14 +460,14 @@ restart:
 						if (curmsg < maxmsg) {
 							deleted[curmsg] = !deleted[curmsg];
 							if (deleted[curmsg]) {
-								if (ast_streamfile(chan, "vm-deleted"))
+								if (ast_streamfile(chan, "vm-deleted", chan->language))
 									goto out;
 							} else {
-								if (ast_streamfile(chan, "vm-undeleted"))
+								if (ast_streamfile(chan, "vm-undeleted", chan->language))
 									goto out;
 							}
 						} else {
-							if (ast_streamfile(chan, "vm-nomore"))
+							if (ast_streamfile(chan, "vm-nomore", chan->language))
 								goto out;
 						}
 						break;
@@ -457,7 +480,7 @@ restart:
 							else if (d < 0)
 								goto out;
 						} else {
-							if (ast_streamfile(chan, "vm-nomore"))
+							if (ast_streamfile(chan, "vm-nomore", chan->language))
 								goto out;
 						}
 						state = STATE_MESSAGE_PLAYING;
@@ -487,7 +510,7 @@ restart:
 							else if (d < 0)
 								goto out;
 						} else {
-							if (ast_streamfile(chan, "vm-nomore"))
+							if (ast_streamfile(chan, "vm-nomore", chan->language))
 								goto out;
 						}
 						state = STATE_MESSAGE_PLAYING;
@@ -495,7 +518,7 @@ restart:
 					/* XXX Message compose? It's easy!  Just read their # and, assuming it's in the config, 
 					       call the routine as if it were called from the PBX proper XXX */
 					case '#':
-						if (ast_streamfile(chan, "vm-goodbye"))
+						if (ast_streamfile(chan, "vm-goodbye", chan->language))
 							goto out;
 						if (ast_waitstream(chan, ""))
 							goto out;

asterisk.c

@@ -3,7 +3,7 @@
  *
  * Top level source file for asterisk
  * 
- * Copyright (C) 1999, Adtran Inc. and Linux Support Services, LLC
+ * Copyright (C) 1999, Mark Spencer
  *
  * Mark Spencer <markster@linux-support.net>
  *
@@ -15,14 +15,28 @@
 #include <stdlib.h>
 #include <asterisk/logger.h>
 #include <asterisk/options.h>
+#include <asterisk/cli.h>
+#include <asterisk/channel.h>
 #include <stdio.h>
 #include <signal.h>
+#include <sched.h>
+#include <pthread.h>
+#include <readline/readline.h>
+#include <readline/history.h>
 #include "asterisk.h"
 
 int option_verbose=0;
 int option_debug=0;
 int option_nofork=0;
 int option_quiet=0;
+int option_console=0;
+int option_highpriority=0;
+int fully_booted = 0;
+
+char defaultlanguage[MAX_LANGUAGE] = DEFAULT_LANGUAGE;
+
+#define HIGH_PRIORITY 1
+#define HIGH_PRIORITY_SCHED SCHED_RR
 
 static void urg_handler(int num)
 {
@@ -30,37 +44,146 @@ static void urg_handler(int num)
 	   system call.  We don't actually need to do anything though.  */
 	if (option_debug)
 		ast_log(LOG_DEBUG, "Urgent handler\n");
+	signal(num, urg_handler);
 	return;
 }
 
+static void set_title(char *text)
+{
+	/* Set an X-term or screen title */
+	if (getenv("TERM") && strstr(getenv("TERM"), "xterm"))
+		fprintf(stdout, "\033]2;%s\007", text);
+}
+
+static void set_icon(char *text)
+{
+	if (getenv("TERM") && strstr(getenv("TERM"), "xterm"))
+		fprintf(stdout, "\033]1;%s\007", text);
+}
+
+static int set_priority(int pri)
+{
+	struct sched_param sched;
+	/* We set ourselves to a high priority, that we might pre-empt everything
+	   else.  If your PBX has heavy activity on it, this is a good thing.  */
+	if (pri) {  
+		sched.sched_priority = HIGH_PRIORITY;
+		if (sched_setscheduler(0, HIGH_PRIORITY_SCHED, &sched)) {
+			ast_log(LOG_WARNING, "Unable to set high priority\n");
+			return -1;
+		}
+	} else {
+		sched.sched_priority = 0;
+		if (sched_setscheduler(0, SCHED_OTHER, &sched)) {
+			ast_log(LOG_WARNING, "Unable to set normal priority\n");
+			return -1;
+		}
+	}
+	return 0;
+}
+
 static void quit_handler(int num)
 {
+	static pthread_mutex_t quitlock = PTHREAD_MUTEX_INITIALIZER;
+	char filename[80] = "";
+	if (getenv("HOME")) 
+		snprintf(filename, sizeof(filename), "%s/.asterisk_history", getenv("HOME"));
+	/* Quit only once */
+	pthread_mutex_lock(&quitlock);
 	/* Called on exit */
 	if (option_verbose)
 		ast_verbose("Asterisk ending (%d).\n", num);
 	else if (option_debug)
 		ast_log(LOG_DEBUG, "Asterisk ending (%d).\n", num);
+	if (strlen(filename))
+		write_history(filename);
 	exit(0);
 }
 
+static pthread_t consolethread = -1;
+
+static void console_verboser(char *s, int pos, int replace, int complete)
+{
+	/* Return to the beginning of the line */
+	if (!pos)
+		fprintf(stdout, "\r");
+	fprintf(stdout, s + pos);
+	fflush(stdout);
+	if (complete)
+	/* Wake up a select()ing console */
+		pthread_kill(consolethread, SIGURG);
+}
+
+static void consolehandler(char *s)
+{
+	/* Called when readline data is available */
+	if (s && strlen(s))
+		add_history(s);
+	/* Give the console access to the shell */
+	if (s) {
+		if (s[0] == '!') {
+			if (s[1])
+				system(s+1);
+			else
+				system(getenv("SHELL") ? getenv("SHELL") : "/bin/sh");
+		} else 
+		ast_cli_command(STDOUT_FILENO, s);
+		if (!strcasecmp(s, "help"))
+			fprintf(stdout, "          !<command>   Executes a given shell command\n");
+	} else
+		fprintf(stdout, "\nUse \"quit\" to exit\n");
+}
+
+static char quit_help[] = 
+"Usage: quit\n"
+"       Exits Asterisk.\n";
+
+static int handle_quit(int fd, int argc, char *argv[])
+{
+	if (argc != 1)
+		return RESULT_SHOWUSAGE;
+	quit_handler(0);
+	return RESULT_SUCCESS;
+}
+
+#define ASTERISK_PROMPT "*CLI> "
+
+static struct ast_cli_entry quit = 	{ { "quit", NULL }, handle_quit, "Exit Asterisk", quit_help };
+
+static char *cli_generator(char *text, int state)
+{
+	return ast_cli_generator(rl_line_buffer, text, state);
+}
+
 int main(int argc, char *argv[])
 {
 	char c;
+	fd_set rfds;
+	int res;
+	char filename[80] = "";
+	if (getenv("HOME")) 
+		snprintf(filename, sizeof(filename), "%s/.asterisk_history", getenv("HOME"));
 	/* Check if we're root */
 	if (geteuid()) {
 		ast_log(LOG_ERROR, "Must be run as root\n");
 		exit(1);
 	}
 	/* Check for options */
-	while((c=getopt(argc, argv, "dvq")) != EOF) {
+	while((c=getopt(argc, argv, "dvqpc")) != EOF) {
 		switch(c) {
 		case 'd':
 			option_debug++;
 			option_nofork++;
-			option_verbose++;
+			break;
+		case 'c':
+			option_console++;
+			option_nofork++;
+		case 'p':
+			option_highpriority++;
 			break;
 		case 'v':
 			option_verbose++;
+			option_nofork++;
 			break;
 		case 'q':
 			option_quiet++;
@@ -69,12 +192,15 @@ int main(int argc, char *argv[])
 			exit(1);
 		}
 	}
+	ast_register_verbose(console_verboser);
 	/* Print a welcome message if desired */
-	if (option_verbose) {
-		ast_verbose( "Asterisk, Copyright (C) 1999 Adtran, Inc. and Linux Support Services, LLC\n");
+	if (option_verbose || option_console) {
+		ast_verbose( "Asterisk, Copyright (C) 1999 Mark Spencer\n");
 		ast_verbose( "Written by Mark Spencer <markster@linux-support.net>\n");
 		ast_verbose( "=========================================================================\n");
 	}
+	if (option_console && !option_verbose) 
+		ast_verbose("[ Booting...");
 	signal(SIGURG, urg_handler);
 	signal(SIGINT, quit_handler);
 	signal(SIGTERM, quit_handler);
@@ -85,10 +211,42 @@ int main(int argc, char *argv[])
 		exit(1);
 	if (load_modules())
 		exit(1);
+	if (set_priority(option_highpriority))
+		exit(1);
 	/* We might have the option of showing a console, but for now just
 	   do nothing... */
-	if (option_verbose)
+	if (option_console && !option_verbose)
+		ast_verbose(" ]\n");
+	if (option_verbose || option_console)
 		ast_verbose( "Asterisk Ready.\n");
-	select(0,NULL,NULL,NULL,NULL);
+	fully_booted = 1;
+	if (option_console) {
+		/* Console stuff now... */
+		/* Register our quit function */
+		char title[256];
+		set_icon("Asterisk");
+		snprintf(title, sizeof(title), "Asterisk Console (pid %d)", getpid());
+		set_title(title);
+	    ast_cli_register(&quit);
+		consolethread = pthread_self();
+		if (strlen(filename))
+			read_history(filename);
+		rl_callback_handler_install(ASTERISK_PROMPT, consolehandler);
+		rl_completion_entry_function = (Function *)cli_generator;
+		for(;;) {
+			FD_ZERO(&rfds);
+			FD_SET(STDIN_FILENO, &rfds);
+			res = select(STDIN_FILENO + 1, &rfds, NULL, NULL, NULL);
+			if (res > 0) {
+				rl_callback_read_char();
+			} else if (res < 1) {
+				rl_forced_update_display();
+			}
+	
+		}	
+	} else {
+		/* Do nothing */
+		select(0,NULL,NULL,NULL,NULL);
+	}
 	return 0;
 }

asterisk.h

@@ -3,7 +3,7 @@
  *
  * General Definitions for Asterisk top level program
  * 
- * Copyright (C) 1999, Adtran Inc. and Linux Support Services, LLC
+ * Copyright (C) 1999, Mark Spencer
  *
  * Mark Spencer <markster@linux-support.net>
  *
@@ -14,6 +14,8 @@
 #ifndef _ASTERISK_H
 #define _ASTERISK_H
 
+#define DEFAULT_LANGUAGE "en"
+
 #define AST_CONFIG_DIR "/etc/asterisk"
 #define AST_MODULE_DIR "/usr/lib/asterisk/modules"
 #define AST_SPOOL_DIR  "/var/spool/asterisk"

audiofile.patch

@@ -0,0 +1,24 @@
+diff -uNr audiofile-0.1.9.old/libaudiofile/audiofile.c audiofile-0.1.9/libaudiofile/audiofile.c
+--- audiofile-0.1.9.old/libaudiofile/audiofile.c	Fri Jul 23 12:57:56 1999
++++ audiofile-0.1.9/libaudiofile/audiofile.c	Fri Dec 10 18:43:30 1999
+@@ -488,6 +488,20 @@
+ 	return afOpenVirtualFile(af_virtual_file_new_for_file(fp), mode, setup);
+ }
+ 
++AFfilehandle afOpenFD(int fd, const char *mode, AFfilesetup setup)
++{
++	FILE			*fp;
++
++	fp = fdopen(fd, mode);
++	if (fp == NULL)
++	{
++		_af_error(AF_BAD_OPEN);
++		return AF_NULL_FILEHANDLE;
++	}
++
++	return afOpenVirtualFile(af_virtual_file_new_for_file(fp), mode, setup);
++}
++
+ int afGetFileFormat (AFfilehandle file, int *version)
+ {
+ 	assert(file);

channel.c

@@ -3,7 +3,7 @@
  *
  * Channel Management
  * 
- * Copyright (C) 1999, Adtran Inc. and Linux Support Services, LLC
+ * Copyright (C) 1999, Mark Spencer
  *
  * Mark Spencer <markster@linux-support.net>
  *
@@ -18,12 +18,35 @@
 #include <sys/time.h>
 #include <signal.h>
 #include <errno.h>
+#include <unistd.h>
 #include <asterisk/sched.h>
 #include <asterisk/options.h>
 #include <asterisk/channel.h>
 #include <asterisk/channel_pvt.h>
 #include <asterisk/logger.h>
 #include <asterisk/file.h>
+#include <asterisk/translate.h>
+
+
+
+#ifdef DEBUG_MUTEX
+/* Convenient mutex debugging functions */
+#define PTHREAD_MUTEX_LOCK(a) __PTHREAD_MUTEX_LOCK(__FUNCTION__, a)
+#define PTHREAD_MUTEX_UNLOCK(a) __PTHREAD_MUTEX_UNLOCK(__FUNCTION__, a)
+
+static int __PTHREAD_MUTEX_LOCK(char *f, pthread_mutex_t *a) {
+	ast_log(LOG_DEBUG, "Locking %p (%s)\n", a, f); 
+	return pthread_mutex_lock(a);
+}
+
+static int __PTHREAD_MUTEX_UNLOCK(char *f, pthread_mutex_t *a) {
+	ast_log(LOG_DEBUG, "Unlocking %p (%s)\n", a, f); 
+	return pthread_mutex_unlock(a);
+}
+#else
+#define PTHREAD_MUTEX_LOCK(a) pthread_mutex_lock(a)
+#define PTHREAD_MUTEX_UNLOCK(a) pthread_mutex_unlock(a)
+#endif
 
 struct chanlist {
 	char type[80];
@@ -32,6 +55,7 @@ struct chanlist {
 	struct ast_channel * (*requester)(char *type, int format, void *data);
 	struct chanlist *next;
 } *backends = NULL;
+struct ast_channel *channels = NULL;
 
 /* Protect the channel list (highly unlikely that two things would change
    it at the same time, but still! */
@@ -42,7 +66,7 @@ int ast_channel_register(char *type, char *description, int capabilities,
 		struct ast_channel *(*requester)(char *type, int format, void *data))
 {
 	struct chanlist *chan, *last=NULL;
-	if (pthread_mutex_lock(&chlock)) {
+	if (PTHREAD_MUTEX_LOCK(&chlock)) {
 		ast_log(LOG_WARNING, "Unable to lock channel list\n");
 		return -1;
 	}
@@ -50,7 +74,7 @@ int ast_channel_register(char *type, char *description, int capabilities,
 	while(chan) {
 		if (!strcasecmp(type, chan->type)) {
 			ast_log(LOG_WARNING, "Already have a handler for type '%s'\n", type);
-			pthread_mutex_unlock(&chlock);
+			PTHREAD_MUTEX_UNLOCK(&chlock);
 			return -1;
 		}
 		last = chan;
@@ -59,7 +83,7 @@ int ast_channel_register(char *type, char *description, int capabilities,
 	chan = malloc(sizeof(struct chanlist));
 	if (!chan) {
 		ast_log(LOG_WARNING, "Out of memory\n");
-		pthread_mutex_unlock(&chlock);
+		PTHREAD_MUTEX_UNLOCK(&chlock);
 		return -1;
 	}
 	strncpy(chan->type, type, sizeof(chan->type));
@@ -75,7 +99,7 @@ int ast_channel_register(char *type, char *description, int capabilities,
 		ast_log(LOG_DEBUG, "Registered handler for '%s' (%s)\n", chan->type, chan->description);
 	else if (option_verbose > 1)
 		ast_verbose( VERBOSE_PREFIX_2 "Registered channel type '%s' (%s)\n", chan->type, chan->description);
-	pthread_mutex_unlock(&chlock);
+	PTHREAD_MUTEX_UNLOCK(&chlock);
 	return 0;
 }
 
@@ -83,6 +107,7 @@ struct ast_channel *ast_channel_alloc(void)
 {
 	struct ast_channel *tmp;
 	struct ast_channel_pvt *pvt;
+	PTHREAD_MUTEX_LOCK(&chlock);
 	tmp = malloc(sizeof(struct ast_channel));
 	memset(tmp, 0, sizeof(struct ast_channel));
 	if (tmp) {
@@ -97,9 +122,15 @@ struct ast_channel *ast_channel_alloc(void)
 				tmp->state = AST_STATE_DOWN;
 				tmp->stack = -1;
 				tmp->streamid = -1;
+				tmp->appl = NULL;
+				tmp->data = NULL;
+				pthread_mutex_init(&tmp->lock, NULL);
 				strncpy(tmp->context, "default", sizeof(tmp->context));
+				strncpy(tmp->language, defaultlanguage, sizeof(tmp->language));
 				strncpy(tmp->exten, "s", sizeof(tmp->exten));
 				tmp->priority=1;
+				tmp->next = channels;
+				channels= tmp;
 			} else {
 				ast_log(LOG_WARNING, "Unable to create schedule context\n");
 				free(tmp);
@@ -112,9 +143,62 @@ struct ast_channel *ast_channel_alloc(void)
 		}
 	} else 
 		ast_log(LOG_WARNING, "Out of memory\n");
+	PTHREAD_MUTEX_UNLOCK(&chlock);
 	return tmp;
 }
 
+struct ast_channel *ast_channel_walk(struct ast_channel *prev)
+{
+	struct ast_channel *l, *ret=NULL;
+	PTHREAD_MUTEX_LOCK(&chlock);
+	l = channels;
+	if (!prev) {
+		PTHREAD_MUTEX_UNLOCK(&chlock);
+		return l;
+	}
+	while(l) {
+		if (l == prev)
+			ret = l->next;
+		l = l->next;
+	}
+	PTHREAD_MUTEX_UNLOCK(&chlock);
+	return ret;
+	
+}
+
+void ast_channel_free(struct ast_channel *chan)
+{
+	struct ast_channel *last=NULL, *cur;
+	PTHREAD_MUTEX_LOCK(&chlock);
+	cur = channels;
+	while(cur) {
+		if (cur == chan) {
+			if (last)
+				last->next = cur->next;
+			else
+				channels = cur->next;
+			break;
+		}
+		last = cur;
+		cur = cur->next;
+	}
+	if (!cur)
+		ast_log(LOG_WARNING, "Unable to find channel in list\n");
+	if (chan->pvt->pvt)
+		ast_log(LOG_WARNING, "Channel '%s' may not have been hung up properly\n", chan->name);
+	if (chan->trans)
+		ast_log(LOG_WARNING, "Hard hangup called on '%s' while a translator is in place!  Expect a failure.\n", chan->name);
+	if (chan->pbx) 
+		ast_log(LOG_WARNING, "PBX may not have been terminated properly on '%s'\n", chan->name);
+	if (chan->dnid)
+		free(chan->dnid);
+	if (chan->callerid)
+		free(chan->callerid);	
+	pthread_mutex_destroy(&chan->lock);
+	free(chan);
+	PTHREAD_MUTEX_UNLOCK(&chlock);
+}
+
 int ast_softhangup(struct ast_channel *chan)
 {
 	int res = 0;
@@ -149,18 +233,7 @@ int ast_hangup(struct ast_channel *chan)
 		ast_log(LOG_DEBUG, "Hanging up channel '%s'\n", chan->name);
 	if (chan->pvt->hangup)
 		res = chan->pvt->hangup(chan);
-	if (chan->pvt->pvt)
-		ast_log(LOG_WARNING, "Channel '%s' may not have been hung up properly\n", chan->name);
-	if (chan->trans)
-		ast_log(LOG_WARNING, "Hard hangup called on '%s' while a translator is in place!  Expect a failure.\n", chan->name);
-	if (chan->pbx) 
-		ast_log(LOG_WARNING, "PBX may not have been terminated properly on '%s'\n", chan->name);
-	if (chan->dnid)
-		free(chan->dnid);
-	if (chan->callerid)
-		free(chan->callerid);	
-	free(chan->pvt);
-	free(chan);
+	ast_channel_free(chan);
 	return res;
 }
 
@@ -169,7 +242,7 @@ void ast_channel_unregister(char *type)
 	struct chanlist *chan, *last=NULL;
 	if (option_debug)
 		ast_log(LOG_DEBUG, "Unregistering channel type '%s'\n", type);
-	if (pthread_mutex_lock(&chlock)) {
+	if (PTHREAD_MUTEX_LOCK(&chlock)) {
 		ast_log(LOG_WARNING, "Unable to lock channel list\n");
 		return;
 	}
@@ -181,13 +254,13 @@ void ast_channel_unregister(char *type)
 			else
 				backends = backends->next;
 			free(chan);
-			pthread_mutex_unlock(&chlock);
+			PTHREAD_MUTEX_UNLOCK(&chlock);
 			return;
 		}
 		last = chan;
 		chan = chan->next;
 	}
-	pthread_mutex_unlock(&chlock);
+	PTHREAD_MUTEX_UNLOCK(&chlock);
 }
 
 int ast_answer(struct ast_channel *chan)
@@ -315,6 +388,16 @@ struct ast_frame *ast_read(struct ast_channel *chan)
 	return f;
 }
 
+int ast_sendtext(struct ast_channel *chan, char *text)
+{
+	int res = 0;
+	CHECK_BLOCKING(chan);
+	if (chan->pvt->send_text)
+		res = chan->pvt->send_text(chan, text);
+	chan->blocking = 0;
+	return res;
+}
+
 int ast_write(struct ast_channel *chan, struct ast_frame *fr)
 {
 	int res = -1;
@@ -341,22 +424,26 @@ struct ast_channel *ast_request(char *type, int format, void *data)
 {
 	struct chanlist *chan;
 	struct ast_channel *c = NULL;
-	if (pthread_mutex_lock(&chlock)) {
+	if (PTHREAD_MUTEX_LOCK(&chlock)) {
 		ast_log(LOG_WARNING, "Unable to lock channel list\n");
 		return NULL;
 	}
 	chan = backends;
 	while(chan) {
 		if (!strcasecmp(type, chan->type)) {
+			if (!(chan->capabilities & format)) {
+				format = ast_translator_best_choice(format, chan->capabilities);
+			}
+			PTHREAD_MUTEX_UNLOCK(&chlock);
 			if (chan->requester)
 				c = chan->requester(type, format, data);
-			pthread_mutex_unlock(&chlock);
-			break;
+			return c;
 		}
 		chan = chan->next;
 	}
 	if (!chan)
 		ast_log(LOG_WARNING, "No channel type registered for '%s'\n", type);
+	PTHREAD_MUTEX_UNLOCK(&chlock);
 	return c;
 }
 
@@ -379,9 +466,10 @@ int ast_readstring(struct ast_channel *c, char *s, int len, int timeout, int fti
 	if (!len)
 		return -1;
 	do {
-		if (c->streamid > -1) {
+		if ((c->streamid > -1) || (c->trans && (c->trans->streamid > -1))) {
 			d = ast_waitstream(c, AST_DIGIT_ANY);
 			ast_stopstream(c);
+			usleep(1000);
 			if (!d)
 				d = ast_waitfordigit(c, to);
 		} else {

channels/DialTone.h

@@ -3,7 +3,7 @@
   *
   * Source: DialTone.ulaw
   *
-  * Copyright (C) 1999, Mark Spencer and Linux Support Services
+  * Copyright (C) 1999, Mark Spencer
   *
   * Distributed under the terms of the GNU General Public License
   *

channels/Makefile

@@ -3,7 +3,7 @@
 # 
 # Makefile for Channel backends (dynamically loaded)
 #
-# Copyright (C) 1999, Adtran Inc. and Linux Support Services, LLC
+# Copyright (C) 1999, Mark Spencer
 #
 # Mark Spencer <markster@linux-support.net>
 #
@@ -11,7 +11,13 @@
 # the GNU General Public License
 #
 
-CHANNEL_LIBS=chan_vofr.so chan_ixj.so
+CHANNEL_LIBS=chan_vofr.so chan_modem.so \
+	     chan_modem_aopen.so chan_iax.so chan_oss.so
+
+CHANNEL_LIBS+=$(shell [ -f /usr/include/linux/ixjuser.h ] && echo chan_phone.so)
+
+CFLAGS+=-Wno-missing-prototypes -Wno-missing-declarations
+CFLAGS+=$(shell [ ! -f /usr/include/linux/if_wanpipe.h ] && echo " -DOLD_SANGOMA_API")
 
 CFLAGS+=#-DVOFRDUMPER
 
@@ -23,5 +29,15 @@ clean:
 %.so : %.o
 	$(CC) -shared -Xlinker -x -o $@ $<
 
+#libiax.a: libiax.o
+#	rm -f libiax.a
+#	$(AR) cr libiax.a libiax.o
+
+#chan_iax.so: chan_iax.o libiax.a
+#	$(CC) -shared -Xlinker -x -o $@ $< -L. -liax
+
+#chan_modem.so : chan_modem.o
+#	$(CC) -rdynamic -shared -Xlinker -x -o $@ $<
+
 install: all
 	for x in $(CHANNEL_LIBS); do $(INSTALL) -m 755 $$x $(MODULES_DIR) ; done

channels/adtranvofr.h

@@ -3,7 +3,7 @@
  *
  * Implementation of Voice over Frame Relay, Adtran Style
  * 
- * Copyright (C) 1999, Adtran Inc. and Linux Support Services, LLC
+ * Copyright (C) 1999, Mark Spencer
  *
  * Mark Spencer <markster@linux-support.net>
  *

channels/chan_modem.c

@@ -0,0 +1,830 @@
+/*
+ * Asterisk -- A telephony toolkit for Linux.
+ *
+ * A/Open ITU-56/2 Voice Modem Driver (Rockwell, IS-101, and others)
+ * 
+ * Copyright (C) 1999, Mark Spencer
+ *
+ * Mark Spencer <markster@linux-support.net>
+ *
+ * This program is free software, distributed under the terms of
+ * the GNU General Public License
+ */
+
+#include <stdio.h>
+#include <pthread.h>
+#include <string.h>
+#include <asterisk/channel.h>
+#include <asterisk/channel_pvt.h>
+#include <asterisk/config.h>
+#include <asterisk/logger.h>
+#include <asterisk/module.h>
+#include <asterisk/pbx.h>
+#include <asterisk/options.h>
+#include <asterisk/vmodem.h>
+#include <sys/socket.h>
+#include <sys/time.h>
+#include <errno.h>
+#include <unistd.h>
+#include <stdlib.h>
+#include <arpa/inet.h>
+#include <fcntl.h>
+#include <sys/ioctl.h>
+#include <sys/termios.h>
+#include <sys/signal.h>
+#include <ctype.h>
+
+/* Up to 10 seconds for an echo to arrive */
+#define ECHO_TIMEOUT 10
+
+static char *desc = "Generic Voice Modem Driver";
+static char *tdesc = "Generic Voice Modem Channel Driver";
+static char *type = "Modem";
+static char *config = "modem.conf";
+static char dialtype = 'T';
+static int gmode = MODEM_MODE_IMMEDIATE;
+
+/* Default modem type */
+static char mtype[80] = "autodetect";
+/* Default context for incoming calls */
+static char context[AST_MAX_EXTENSION]= "default";
+
+/* Default language */
+static char language[MAX_LANGUAGE] = "";
+
+/* Initialization String */
+static char initstr[AST_MAX_INIT_STR] = "ATE1Q0";
+
+static int usecnt =0;
+
+static int baudrate = 115200;
+
+static int stripmsd = 0;
+
+static pthread_mutex_t usecnt_lock = PTHREAD_MUTEX_INITIALIZER;
+
+/* Protect the interface list (of ast_modem_pvt's) */
+static pthread_mutex_t iflock = PTHREAD_MUTEX_INITIALIZER;
+
+/* Protect the monitoring thread, so only one process can kill or start it, and not
+   when it's doing something critical. */
+static pthread_mutex_t monlock = PTHREAD_MUTEX_INITIALIZER;
+
+/* This is the thread for the monitor which checks for input on the channels
+   which are not currently in use.  */
+static pthread_t monitor_thread = -1;
+
+static int restart_monitor(void);
+
+/* The private structures of the Phone Jack channels are linked for
+   selecting outgoing channels */
+   
+static struct ast_modem_pvt  *iflist = NULL;
+
+static int modem_digit(struct ast_channel *ast, char digit)
+{
+	struct ast_modem_pvt *p;
+	p = ast->pvt->pvt;
+	if (p->mc->dialdigit)
+		return p->mc->dialdigit(p, digit);
+	else ast_log(LOG_DEBUG, "Channel %s lacks digit dialing\n");
+	return 0;
+}
+
+static struct ast_modem_driver *drivers = NULL;
+
+static struct ast_frame *modem_read(struct ast_channel *);
+
+static struct ast_modem_driver *find_capability(char *ident)
+{
+	struct ast_modem_driver *mc;
+	int x;
+	mc = drivers;
+	while(mc) {
+		for (x=0;mc->idents[x];x++) {
+			if (!strcmp(ident, mc->idents[x])) 
+				break;
+		}
+		if (mc->idents[x])
+			break;
+		mc = mc->next;
+	}
+	if (mc) {
+		if (mc->incusecnt)
+			mc->incusecnt();
+	}
+	return mc;
+}
+
+static struct ast_modem_driver *find_driver(char *drv)
+{
+	struct ast_modem_driver *mc;
+	mc = drivers;
+	while(mc) {
+		if (!strcasecmp(mc->name, drv))	
+			break;
+		mc = mc->next;
+	}
+	if (mc) {
+		if (mc->incusecnt)
+			mc->incusecnt();
+	}
+	return mc;
+}
+
+int ast_register_modem_driver(struct ast_modem_driver *mc)
+{
+	mc->next = drivers;
+	drivers = mc;
+	return 0;
+}
+
+int ast_unregister_modem_driver(struct ast_modem_driver *mc)
+{
+	struct ast_modem_driver *last = NULL, *cur;
+	cur = drivers;
+	while(cur) {
+		if (cur == mc) {
+			if (last)
+				last->next = mc->next;
+			else
+				drivers = mc->next;
+			return 0;
+		}
+		cur = cur->next;
+	}
+	return -1;
+}
+
+static int modem_call(struct ast_channel *ast, char *idest, int timeout)
+{
+	struct ast_modem_pvt *p;
+	int ms = timeout;
+	char rdest[80], *where;
+	strncpy(rdest, idest, sizeof(rdest));
+	strtok(rdest, ":");
+	where = strtok(NULL, ":");
+	if (!where) {
+		ast_log(LOG_WARNING, "Destination %s requres a real destination (device:destination)\n", idest);
+		return -1;
+	}
+	p = ast->pvt->pvt;
+	if ((ast->state != AST_STATE_DOWN) && (ast->state != AST_STATE_RESERVED)) {
+		ast_log(LOG_WARNING, "modem_call called on %s, neither down nor reserved\n", ast->name);
+		return -1;
+	}
+	if (!strcasecmp(where, "handset")) {
+		if (p->mc->setdev)
+			if (p->mc->setdev(p, MODEM_DEV_HANDSET))
+				return -1;
+		/* Should be immediately up */
+		ast->state = AST_STATE_UP;
+	} else {
+		if (p->mc->setdev)
+			if (p->mc->setdev(p, MODEM_DEV_TELCO_SPK))
+				return -1;
+		if (p->mc->dial)
+			p->mc->dial(p, where + p->stripmsd);
+		ast->state = AST_STATE_DIALING;
+		while((ast->state != AST_STATE_UP) && (ms > 0)) {
+			ms = ast_waitfor(ast, ms);
+			/* Just read packets and watch what happens */
+			if (ms > 0) {
+				if (!modem_read(ast))
+					return -1;
+			}
+		}
+		if (ms < 0)	
+			return -1;
+	}
+	return 0;
+}
+
+int ast_modem_send(struct ast_modem_pvt *p, char *cmd, int len)
+{
+	int res;
+	if (!len) {
+		fprintf(p->f, "%s\r\n", cmd);
+		res = ast_modem_expect(p, cmd, ECHO_TIMEOUT);
+		if (res) {
+			ast_log(LOG_WARNING, "Unexpected reply %s\n", p->response);
+			return -1;
+		}
+		return 0;
+	} else {
+		if (fwrite(cmd, 1, len, p->f) < len)
+			return -1;
+		return 0;
+	}
+}
+
+int ast_modem_read_response(struct ast_modem_pvt *p, int timeout)
+{
+	int res = -1;
+	timeout *= 1000;
+	strncpy(p->response, "(No Response)", sizeof(p->response));
+	do {
+		res = ast_waitfor_n_fd(&p->fd, 1, &timeout);
+		if (res < 0) {
+			return -1;
+		}
+		/* Read a response */
+		fgets(p->response, sizeof(p->response), p->f);
+		return 0;
+	} while(timeout > 0);
+	return -1;
+}
+
+int ast_modem_expect(struct ast_modem_pvt *p, char *result, int timeout)
+{
+	int res = -1;
+	timeout *= 1000;
+	strncpy(p->response, "(No Response)", sizeof(p->response));
+	do {
+		res = ast_waitfor_n_fd(&p->fd, 1, &timeout);
+		if (res < 0) {
+			return -1;
+		}
+		/* Read a response */
+		fgets(p->response, sizeof(p->response), p->f);
+#if 0
+		fprintf(stderr, "Modem said: %s", p->response);
+#endif
+		if (!strncasecmp(p->response, result, strlen(result))) 
+			return 0;
+	} while(timeout > 0);
+	return -1;
+}
+
+void ast_modem_trim(char *s)
+{
+	int x;
+	x = strlen(s) - 1;
+	while(x >= 0) {
+		if ((s[x] != '\r') && (s[x] != '\n') && (s[x] != ' '))
+			break;
+		s[x] = '\0';
+		x--;
+	}
+}
+
+static int modem_setup(struct ast_modem_pvt *p, int baudrate)
+{
+	/* Make sure there's a modem there and that it's in a reasonable 
+	   mode.  Set the baud rate, etc.  */
+	char identity[256];
+	char *ident = NULL;
+	char etx[2] = { 0x10, 0x03 };
+	if (option_debug)
+		ast_log(LOG_DEBUG, "Setting up modem %s\n", p->dev);
+	if (ast_modem_send(p, etx, 2)) {
+		ast_log(LOG_WARNING, "Failed to send ETX?\n");
+		return -1;
+	}
+	if (ast_modem_send(p, "\r\n", 2)) {
+		ast_log(LOG_WARNING, "Failed to send enter?\n");
+		return -1;
+	}
+	/* Read any outstanding stuff */
+	while(!ast_modem_read_response(p, 0));
+	if (ast_modem_send(p, "ATZ", 0)) {
+		ast_log(LOG_WARNING, "Modem not responding on %s\n", p->dev);
+		return -1;
+	}
+	if (ast_modem_expect(p, "OK", ECHO_TIMEOUT)) {
+		ast_log(LOG_WARNING, "Modem reset failed: %s\n", p->response);
+		return -1;
+	}
+	if (ast_modem_send(p, p->initstr, 0)) {
+		ast_log(LOG_WARNING, "Modem not responding on %s\n", p->dev);
+		return -1;
+	}
+	if (ast_modem_expect(p, "OK", ECHO_TIMEOUT)) {
+		ast_log(LOG_WARNING, "Modem initialization failed: %s\n", p->response);
+		return -1;
+	}
+	if (ast_modem_send(p, "ATI3", 0)) {
+		ast_log(LOG_WARNING, "Modem not responding on %s\n", p->dev);
+		return -1;
+	}
+	if (ast_modem_read_response(p, ECHO_TIMEOUT)) {
+		ast_log(LOG_WARNING, "Modem did not provide identification\n");
+		return -1;
+	}
+	strncpy(identity, p->response, sizeof(identity));
+	ast_modem_trim(identity);
+	if (ast_modem_expect(p, "OK", ECHO_TIMEOUT)) {
+		ast_log(LOG_WARNING, "Modem did not provide identification\n");
+		return -1;
+	}
+	if (!strcasecmp(mtype, "autodetect")) {
+		p->mc = find_capability(identity);
+		if (!p->mc) {
+			ast_log(LOG_WARNING, "Unable to autodetect modem.  You'll need to specify a driver in modem.conf.  Please report modem identification (%s) and which driver works to markster@linux-support.net.\n", identity); 
+			return -1;
+		}
+	} else {
+		p->mc = find_driver(mtype);
+		if (!p->mc) {
+			ast_log(LOG_WARNING, "No driver for modem type '%s'\n", mtype);
+			return -1;
+		}
+	}
+	if (p->mc->init) {
+		if (p->mc->init(p)) {
+			ast_log(LOG_WARNING, "Modem Initialization Failed on '%s', driver %s.\n", p->dev, p->mc->name);
+			p->mc->decusecnt();
+			return -1;
+		}
+	}			
+	if (option_verbose > 2) {
+		ast_verbose(VERBOSE_PREFIX_3 "Configured modem %s with driver %s (%s)\n", p->dev, p->mc->name, p->mc->identify ? (ident = p->mc->identify(p)) : "No identification");
+	}
+	if (ident)
+		free(ident);
+	return 0;
+}
+
+static int modem_hangup(struct ast_channel *ast)
+{
+	struct ast_modem_pvt *p;
+	ast_log(LOG_DEBUG, "modem_hangup(%s)\n", ast->name);
+	p = ast->pvt->pvt;
+	/* Hang up */
+	if (p->mc->hangup)
+		p->mc->hangup(p);
+	/* Re-initialize */
+	if (p->mc->init)
+		p->mc->init(p);
+	ast->state = AST_STATE_DOWN;
+	memset(p->cid, 0, sizeof(p->cid));
+	((struct ast_modem_pvt *)(ast->pvt->pvt))->owner = NULL;
+	pthread_mutex_lock(&usecnt_lock);
+	usecnt--;
+	if (usecnt < 0) 
+		ast_log(LOG_WARNING, "Usecnt < 0???\n");
+	pthread_mutex_unlock(&usecnt_lock);
+	ast_update_use_count();
+	if (option_verbose > 2) 
+		ast_verbose( VERBOSE_PREFIX_3 "Hungup '%s'\n", ast->name);
+	ast->pvt->pvt = NULL;
+	ast->state = AST_STATE_DOWN;
+	restart_monitor();
+	return 0;
+}
+
+static int modem_answer(struct ast_channel *ast)
+{
+	struct ast_modem_pvt *p;
+	int res=0;
+	ast_log(LOG_DEBUG, "modem_hangup(%s)\n", ast->name);
+	p = ast->pvt->pvt;
+	if (p->mc->answer) {
+		res = p->mc->answer(p);
+	}
+	if (!res) {
+		ast->rings = 0;
+		ast->state = AST_STATE_UP;
+	}
+	return res;
+}
+
+static char modem_2digit(char c)
+{
+	if (c == 12)
+		return '#';
+	else if (c == 11)
+		return '*';
+	else if ((c < 10) && (c >= 0))
+		return '0' + c - 1;
+	else
+		return '?';
+}
+
+static struct ast_frame *modem_read(struct ast_channel *ast)
+{
+	struct ast_modem_pvt *p = ast->pvt->pvt;
+	struct ast_frame *fr=NULL;
+	if (p->mc->read)
+		fr = p->mc->read(p);
+	return fr;
+}
+
+static int modem_write(struct ast_channel *ast, struct ast_frame *frame)
+{
+	int res=0;
+	struct ast_modem_pvt *p = ast->pvt->pvt;
+	if (p->mc->write)
+		res = p->mc->write(p, frame);
+	return 0;
+}
+
+struct ast_channel *ast_modem_new(struct ast_modem_pvt *i, int state)
+{
+	struct ast_channel *tmp;
+	tmp = ast_channel_alloc();
+	if (tmp) {
+		snprintf(tmp->name, sizeof(tmp->name), "Modem[%s]/%s", i->mc->name, i->dev + 5);
+		tmp->type = type;
+		tmp->fd = i->fd;
+		tmp->format = i->mc->formats;
+		tmp->state = state;
+		if (state == AST_STATE_RING)
+			tmp->rings = 1;
+		tmp->pvt->pvt = i;
+		tmp->pvt->send_digit = modem_digit;
+		tmp->pvt->call = modem_call;
+		tmp->pvt->hangup = modem_hangup;
+		tmp->pvt->answer = modem_answer;
+		tmp->pvt->read = modem_read;
+		tmp->pvt->write = modem_write;
+		strncpy(tmp->context, i->context, sizeof(tmp->context));
+		if (strlen(i->cid))
+			tmp->callerid = strdup(i->cid);
+		if (strlen(i->language))
+			strncpy(tmp->language,i->language, sizeof(tmp->language));
+		i->owner = tmp;
+		pthread_mutex_lock(&usecnt_lock);
+		usecnt++;
+		pthread_mutex_unlock(&usecnt_lock);
+		ast_update_use_count();
+		if (state != AST_STATE_DOWN) {
+			if (ast_pbx_start(tmp)) {
+				ast_log(LOG_WARNING, "Unable to start PBX on %s\n", tmp->name);
+				ast_hangup(tmp);
+				tmp = NULL;
+			}
+		}
+	} else
+		ast_log(LOG_WARNING, "Unable to allocate channel structure\n");
+	return tmp;
+}
+
+static void modem_mini_packet(struct ast_modem_pvt *i)
+{
+	struct ast_frame *fr;
+	fr = i->mc->read(i);
+	if (fr->frametype == AST_FRAME_CONTROL) {
+		if (fr->subclass == AST_CONTROL_RING) {
+			ast_modem_new(i, AST_STATE_RING);
+			restart_monitor();
+		}
+	}
+}
+
+static void *do_monitor(void *data)
+{
+	fd_set rfds, efds;
+	int n, res;
+	struct ast_modem_pvt *i;
+	/* This thread monitors all the frame relay interfaces which are not yet in use
+	   (and thus do not have a separate thread) indefinitely */
+	/* From here on out, we die whenever asked */
+#if 0
+	if (pthread_setcanceltype(PTHREAD_CANCEL_ASYNCHRONOUS, NULL)) {
+		ast_log(LOG_WARNING, "Unable to set cancel type to asynchronous\n");
+		return NULL;
+	}
+#endif
+	for(;;) {
+		/* Don't let anybody kill us right away.  Nobody should lock the interface list
+		   and wait for the monitor list, but the other way around is okay. */
+		if (pthread_mutex_lock(&monlock)) {
+			ast_log(LOG_ERROR, "Unable to grab monitor lock\n");
+			return NULL;
+		}
+		/* Lock the interface list */
+		if (pthread_mutex_lock(&iflock)) {
+			ast_log(LOG_ERROR, "Unable to grab interface lock\n");
+			pthread_mutex_unlock(&monlock);
+			return NULL;
+		}
+		/* Build the stuff we're going to select on, that is the socket of every
+		   ast_modem_pvt that does not have an associated owner channel */
+		n = -1;
+		FD_ZERO(&rfds);
+		FD_ZERO(&efds);
+		i = iflist;
+		while(i) {
+			if (FD_ISSET(i->fd, &rfds)) 
+				ast_log(LOG_WARNING, "Descriptor %d appears twice (%s)?\n", i->fd, i->dev);
+			if (!i->owner) {
+				/* This needs to be watched, as it lacks an owner */
+				FD_SET(i->fd, &rfds);
+				FD_SET(i->fd, &efds);
+				if (i->fd > n)
+					n = i->fd;
+			}
+			
+			i = i->next;
+		}
+		/* Okay, now that we know what to do, release the interface lock */
+		pthread_mutex_unlock(&iflock);
+		
+		/* And from now on, we're okay to be killed, so release the monitor lock as well */
+		pthread_mutex_unlock(&monlock);
+#if 0
+		ast_log(LOG_DEBUG, "In monitor, n=%d, pid=%d\n", n, getpid());
+#endif
+		/* Wait indefinitely for something to happen */
+		pthread_testcancel();
+		res = select(n + 1, &rfds, NULL, &efds, NULL);
+		pthread_testcancel();
+		/* Okay, select has finished.  Let's see what happened.  */
+		if (res < 1) {
+			ast_log(LOG_WARNING, "select return %d: %s\n", res, strerror(errno));
+			continue;
+		}
+		/* Alright, lock the interface list again, and let's look and see what has
+		   happened */
+		if (pthread_mutex_lock(&iflock)) {
+			ast_log(LOG_WARNING, "Unable to lock the interface list\n");
+			continue;
+		}
+		i = iflist;
+		while(i) {
+			if (FD_ISSET(i->fd, &rfds) || FD_ISSET(i->fd, &efds)) {
+				if (i->owner) {
+					ast_log(LOG_WARNING, "Whoa....  I'm owned but found (%d, %s)...\n", i->fd, i->dev);
+					i = i->next;
+					continue;
+				}
+				modem_mini_packet(i);
+			}
+			i=i->next;
+		}
+		pthread_mutex_unlock(&iflock);
+	}
+	/* Never reached */
+	return NULL;
+	
+}
+
+static int restart_monitor()
+{
+	/* If we're supposed to be stopped -- stay stopped */
+	if (monitor_thread == -2)
+		return 0;
+	if (pthread_mutex_lock(&monlock)) {
+		ast_log(LOG_WARNING, "Unable to lock monitor\n");
+		return -1;
+	}
+	if (monitor_thread == pthread_self()) {
+		pthread_mutex_unlock(&monlock);
+		ast_log(LOG_WARNING, "Cannot kill myself\n");
+		return -1;
+	}
+	if (monitor_thread != -1) {
+		pthread_cancel(monitor_thread);
+		/* Nudge it a little, as it's probably stuck in select */
+		pthread_kill(monitor_thread, SIGURG);
+		pthread_join(monitor_thread, NULL);
+	}
+	/* Start a new monitor */
+	if (pthread_create(&monitor_thread, NULL, do_monitor, NULL) < 0) {
+		pthread_mutex_unlock(&monlock);
+		ast_log(LOG_ERROR, "Unable to start monitor thread.\n");
+		return -1;
+	}
+	pthread_mutex_unlock(&monlock);
+	return 0;
+}
+
+static struct ast_modem_pvt *mkif(char *iface)
+{
+	/* Make a ast_modem_pvt structure for this interface */
+	struct ast_modem_pvt *tmp;
+#if 0
+	int flags;	
+#endif
+	
+	tmp = malloc(sizeof(struct ast_modem_pvt));
+	if (tmp) {
+		tmp->fd = open(iface, O_RDWR | O_NONBLOCK);
+		if (tmp->fd < 0) {
+			ast_log(LOG_WARNING, "Unable to open '%s'\n", iface);
+			free(tmp);
+			return NULL;
+		}
+		strncpy(tmp->language, language, sizeof(tmp->language));
+		tmp->f = fdopen(tmp->fd, "w+");
+		/* Disable buffering */
+		setvbuf(tmp->f, NULL, _IONBF,0);
+		if (tmp->f < 0) {
+			ast_log(LOG_WARNING, "Unable to fdopen '%s'\n", iface);
+			free(tmp);
+			return NULL;
+		}
+#if 0
+		flags = fcntl(tmp->fd, F_GETFL);
+		fcntl(tmp->fd, F_SETFL, flags | O_NONBLOCK);
+#endif
+		tmp->owner = NULL;
+		tmp->ministate = 0;
+		tmp->stripmsd = stripmsd;
+		tmp->dialtype = dialtype;
+		tmp->mode = gmode;
+		memset(tmp->cid, 0, sizeof(tmp->cid));
+		strncpy(tmp->dev, iface, sizeof(tmp->dev));
+		strncpy(tmp->context, context, sizeof(tmp->context));
+		strncpy(tmp->initstr, initstr, sizeof(tmp->initstr));
+		tmp->next = NULL;
+		tmp->obuflen = 0;
+		
+		if (modem_setup(tmp, baudrate) < 0) {
+			ast_log(LOG_WARNING, "Unable to configure modem '%s'\n", iface);
+			free(tmp);
+			return NULL;
+		}
+	}
+	return tmp;
+}
+
+static struct ast_channel *modem_request(char *type, int format, void *data)
+{
+	int oldformat;
+	struct ast_modem_pvt *p;
+	struct ast_channel *tmp = NULL;
+	char dev[80];
+	strncpy(dev, (char *)data, sizeof(dev));
+	strtok(dev, ":");
+	oldformat = format;
+	/* Search for an unowned channel */
+	if (pthread_mutex_lock(&iflock)) {
+		ast_log(LOG_ERROR, "Unable to lock interface list???\n");
+		return NULL;
+	}
+	p = iflist;
+	while(p) {
+		if (!strcmp(dev, p->dev + 5)) {
+			if (p->mc->formats & format) {
+				if (!p->owner) {
+					tmp = ast_modem_new(p, AST_STATE_DOWN);
+					restart_monitor();
+					break;
+				} else
+					ast_log(LOG_WARNING, "Device '%s' is busy\n", p->dev);
+			} else 
+				ast_log(LOG_WARNING, "Asked for a format %d line on %s\n", format, p->dev);
+			break;
+		}
+		p = p->next;
+	}
+	if (!p) 
+		ast_log(LOG_WARNING, "Requested device '%s' does not exist\n", p->dev);
+	
+	pthread_mutex_unlock(&iflock);
+	return tmp;
+}
+
+int load_module()
+{
+	struct ast_config *cfg;
+	struct ast_variable *v;
+	struct ast_modem_pvt *tmp;
+	char driver[80];
+	cfg = ast_load(config);
+
+	/* We *must* have a config file otherwise stop immediately */
+	if (!cfg) {
+		ast_log(LOG_ERROR, "Unable to load config %s\n", config);
+		return -1;
+	}
+	if (pthread_mutex_lock(&iflock)) {
+		/* It's a little silly to lock it, but we mind as well just to be sure */
+		ast_log(LOG_ERROR, "Unable to lock interface list???\n");
+		return -1;
+	}
+	v = ast_variable_browse(cfg, "interfaces");
+	while(v) {
+		/* Create the interface list */
+		if (!strcasecmp(v->name, "device")) {
+				tmp = mkif(v->value);
+				if (tmp) {
+					tmp->next = iflist;
+					iflist = tmp;
+					
+				} else {
+					ast_log(LOG_ERROR, "Unable to register channel '%s'\n", v->value);
+					ast_destroy(cfg);
+					pthread_mutex_unlock(&iflock);
+					unload_module();
+					pthread_mutex_unlock(&iflock);
+					return -1;
+				}
+		} else if (!strcasecmp(v->name, "driver")) {
+			snprintf(driver, sizeof(driver), "chan_modem_%s.so", v->value);
+			if (option_verbose > 1) 
+				ast_verbose(VERBOSE_PREFIX_2 "Loading modem driver %s", driver);
+				
+			if (ast_load_resource(driver)) {
+				ast_log(LOG_ERROR, "Failed to load driver %s\n", driver);
+				ast_destroy(cfg);
+				pthread_mutex_unlock(&iflock);
+				unload_module();
+				return -1;
+			}
+		} else if (!strcasecmp(v->name, "mode")) {
+			if (!strncasecmp(v->value, "ri", 2)) 
+				gmode = MODEM_MODE_WAIT_RING;
+			else if (!strncasecmp(v->value, "im", 2))
+				gmode = MODEM_MODE_IMMEDIATE;
+			else if (!strncasecmp(v->value, "an", 2))
+				gmode = MODEM_MODE_WAIT_ANSWER;
+			else
+				ast_log(LOG_WARNING, "Unknown mode: %s\n", v->value);
+		} else if (!strcasecmp(v->name, "stripmsd")) {
+			stripmsd = atoi(v->value);
+		} else if (!strcasecmp(v->name, "type")) {
+			strncpy(mtype, v->value, sizeof(mtype));
+		} else if (!strcasecmp(v->name, "initstr")) {
+			strncpy(initstr, v->value, sizeof(initstr));
+		} else if (!strcasecmp(v->name, "dialtype")) {
+			dialtype = toupper(v->value[0]);
+		} else if (!strcasecmp(v->name, "context")) {
+			strncpy(context, v->value, sizeof(context));
+		} else if (!strcasecmp(v->name, "language")) {
+			strncpy(language, v->value, sizeof(language));
+		}
+		v = v->next;
+	}
+	pthread_mutex_unlock(&iflock);
+	if (ast_channel_register(type, tdesc, /* XXX Don't know our types -- maybe we should register more than one XXX */ AST_FORMAT_SLINEAR, modem_request)) {
+		ast_log(LOG_ERROR, "Unable to register channel class %s\n", type);
+		ast_destroy(cfg);
+		unload_module();
+		return -1;
+	}
+	ast_destroy(cfg);
+	/* And start the monitor for the first time */
+	restart_monitor();
+	return 0;
+}
+
+int unload_module()
+{
+	struct ast_modem_pvt *p, *pl;
+	/* First, take us out of the channel loop */
+	ast_channel_unregister(type);
+	if (!pthread_mutex_lock(&iflock)) {
+		/* Hangup all interfaces if they have an owner */
+		p = iflist;
+		while(p) {
+			if (p->owner)
+				ast_softhangup(p->owner);
+			p = p->next;
+		}
+		iflist = NULL;
+		pthread_mutex_unlock(&iflock);
+	} else {
+		ast_log(LOG_WARNING, "Unable to lock the monitor\n");
+		return -1;
+	}
+	if (!pthread_mutex_lock(&monlock)) {
+		if (monitor_thread > -1) {
+			pthread_cancel(monitor_thread);
+			pthread_join(monitor_thread, NULL);
+		}
+		monitor_thread = -2;
+		pthread_mutex_unlock(&monlock);
+	} else {
+		ast_log(LOG_WARNING, "Unable to lock the monitor\n");
+		return -1;
+	}
+
+	if (!pthread_mutex_lock(&iflock)) {
+		/* Destroy all the interfaces and free their memory */
+		p = iflist;
+		while(p) {
+			/* Close the socket, assuming it's real */
+			if (p->fd > -1)
+				close(p->fd);
+			pl = p;
+			p = p->next;
+			/* Free associated memory */
+			free(pl);
+		}
+		iflist = NULL;
+		pthread_mutex_unlock(&iflock);
+	} else {
+		ast_log(LOG_WARNING, "Unable to lock the monitor\n");
+		return -1;
+	}
+		
+	return 0;
+}
+
+int usecount(void)
+{
+	int res;
+	pthread_mutex_lock(&usecnt_lock);
+	res = usecnt;
+	pthread_mutex_unlock(&usecnt_lock);
+	return res;
+}
+
+char *description()
+{
+	return desc;
+}
+

channels/chan_modem_aopen.c

@@ -0,0 +1,480 @@
+/*
+ * Asterisk -- A telephony toolkit for Linux.
+ *
+ * A/Open ITU-56/2 Voice Modem Driver (Rockwell, IS-101, and others)
+ * 
+ * Copyright (C) 1999, Mark Spencer
+ *
+ * Mark Spencer <markster@linux-support.net>
+ *
+ * This program is free software, distributed under the terms of
+ * the GNU General Public License
+ */
+
+#include <stdio.h>
+
+#include <string.h>
+#include <stdlib.h>
+#include <errno.h>
+#include <pthread.h>
+#include <asterisk/vmodem.h>
+#include <asterisk/module.h>
+#include <asterisk/frame.h>
+#include <asterisk/logger.h>
+#include <asterisk/options.h>
+
+#define STATE_COMMAND 	0
+#define STATE_VOICE 	1
+
+#define VRA "40"			/* Number of 100ms of non-ring after a ring cadence after which we consider the lien to be answered */
+#define VRN "100"			/* Number of 100ms of non-ring with no cadence after which we assume an answer */
+
+static char *breakcmd = "\0x10\0x03";
+
+static char *desc = "A/Open (Rockwell Chipset) ITU-2 VoiceModem Driver";
+
+int usecnt;
+pthread_mutex_t usecnt_lock = PTHREAD_MUTEX_INITIALIZER;
+
+static char *aopen_idents[] = {
+	/* Identify A/Open Modem */
+	"V2.210-V90_2M_DLP",
+	NULL
+};
+
+static int aopen_setdev(struct ast_modem_pvt *p, int dev)
+{
+	char cmd[80];
+	if (ast_modem_send(p, "AT#VLS?", 0)) {
+		ast_log(LOG_WARNING, "Unable to select current mode %d\n", dev);
+		return -1;
+	}
+	if (ast_modem_read_response(p, 5)) {
+		ast_log(LOG_WARNING, "Unable to select device %d\n", dev);
+		return -1;
+	}
+	ast_modem_trim(p->response);
+	strncpy(cmd, p->response, sizeof(cmd));
+	if (ast_modem_expect(p, "OK", 5)) {
+		ast_log(LOG_WARNING, "Modem did not respond properly\n");
+		return -1;
+	}
+	if (dev == atoi(cmd)) {
+		/* We're already in the right mode, don't bother changing for fear of
+		   hanging up */
+		return 0;
+	}
+	snprintf(cmd, sizeof(cmd), "AT#VLS=%d", dev);
+	if (ast_modem_send(p, cmd, 0))  {
+		ast_log(LOG_WARNING, "Unable to select device %d\n", dev);
+		return -1;
+	}
+	if (ast_modem_read_response(p, 5)) {
+		ast_log(LOG_WARNING, "Unable to select device %d\n", dev);
+		return -1;
+	}
+	ast_modem_trim(p->response);
+	if (strcasecmp(p->response, "VCON") && strcasecmp(p->response, "OK")) {
+		ast_log(LOG_WARNING, "Unexpected reply: %s\n", p->response);
+		return -1;
+	}
+	return 0;
+}
+
+static int aopen_startrec(struct ast_modem_pvt *p)
+{
+	if (ast_modem_send(p, "AT#VRX", 0) ||
+	     ast_modem_expect(p, "CONNECT", 5)) {
+		ast_log(LOG_WARNING, "Unable to start recording\n");
+		return -1;
+	}
+	p->ministate = STATE_VOICE;
+	return 0;
+}
+
+static int aopen_break(struct ast_modem_pvt *p)
+{
+	if (ast_modem_send(p, "\r\n", 2)) {
+		ast_log(LOG_WARNING, "Failed to send enter?\n");
+		return -1;
+	}
+	if (ast_modem_send(p, breakcmd, 2)) {
+		ast_log(LOG_WARNING, "Failed to break\n");
+		return -1;
+	}
+	if (ast_modem_send(p, "\r\n", 2)) {
+		ast_log(LOG_WARNING, "Failed to send enter?\n");
+		return -1;
+	}
+	/* Read any outstanding junk */
+	while(!ast_modem_read_response(p, 1));
+	if (ast_modem_send(p, "AT", 0)) {
+		/* Modem might be stuck in some weird mode, try to get it out */
+		ast_modem_send(p, "+++", 3);
+		if (ast_modem_expect(p, "OK", 10)) {
+			ast_log(LOG_WARNING, "Modem is not responding\n");
+			return -1;
+		}
+		if (ast_modem_send(p, "AT", 0)) {
+			ast_log(LOG_WARNING, "Modem is not responding\n");
+			return -1;
+		}
+	}
+	if (ast_modem_expect(p, "OK", 5)) {
+		ast_log(LOG_WARNING, "Modem did not respond properly\n");
+		return -1;
+	}
+	return 0;
+}
+
+static int aopen_init(struct ast_modem_pvt *p)
+{
+	if (option_debug)
+		ast_log(LOG_DEBUG, "aopen_init()\n");
+	if (aopen_break(p))
+		return -1;
+	/* Force into command mode */
+	p->ministate = STATE_COMMAND;
+	if (ast_modem_send(p, "AT#BDR=0", 0) ||
+	     ast_modem_expect(p, "OK", 5)) {
+		ast_log(LOG_WARNING, "Unable to set to auto-baud\n");
+		return -1;
+	}
+	if (ast_modem_send(p, "AT#CLS=8", 0) ||
+	     ast_modem_expect(p, "OK", 5)) {
+		ast_log(LOG_WARNING, "Unable to set to voice mode\n");
+		return -1;
+	}
+	if (ast_modem_send(p, "AT#VBS=8", 0) ||
+	     ast_modem_expect(p, "OK", 5)) {
+		ast_log(LOG_WARNING, "Unable to set to 8-bit mode\n");
+		return -1;
+	}
+	if (ast_modem_send(p, "AT#VSR=8000", 0) ||
+	     ast_modem_expect(p, "OK", 5)) {
+		ast_log(LOG_WARNING, "Unable to set to 8000 Hz sampling\n");
+		return -1;
+	}
+	if (ast_modem_send(p, "AT#VLS=0", 0) ||
+	     ast_modem_expect(p, "OK", 5)) {
+		ast_log(LOG_WARNING, "Unable to set to telco interface\n");
+		return -1;
+	}
+	if (ast_modem_send(p, "AT#VRA=" VRA, 0) ||
+	     ast_modem_expect(p, "OK", 5)) {
+		ast_log(LOG_WARNING, "Unable to set to 'ringback goes away' timer\n");
+		return -1;
+	}
+	if (ast_modem_send(p, "AT#VRN=" VRN, 0) ||
+	     ast_modem_expect(p, "OK", 5)) {
+		ast_log(LOG_WARNING, "Unable to set to 'ringback never came timer'\n");
+		return -1;
+	}
+	if (ast_modem_send(p, "AT#VTD=3F,3F,3F", 0) ||
+	     ast_modem_expect(p, "OK", 5)) {
+		ast_log(LOG_WARNING, "Unable to set to tone detection\n");
+		return -1;
+	}
+	
+	return 0;
+}
+
+static struct ast_frame *aopen_handle_escape(struct ast_modem_pvt *p, char esc)
+{
+	/* Handle escaped characters -- but sometimes we call it directly as 
+	   a quick way to cause known responses */
+	p->fr.frametype = AST_FRAME_NULL;
+	p->fr.subclass = 0;
+	p->fr.data = NULL;
+	p->fr.datalen = 0;
+	p->fr.timelen = 0;
+	p->fr.offset = 0;
+	p->fr.mallocd = 0;
+	if (esc)
+		ast_log(LOG_DEBUG, "Escaped character '%c'\n", esc);
+	
+	switch(esc) {
+	case 'R': /* Pseudo ring */
+		p->fr.frametype = AST_FRAME_CONTROL;
+		p->fr.subclass = AST_CONTROL_RING;
+		return &p->fr;
+	case 'X': /* Pseudo connect */
+		p->fr.frametype = AST_FRAME_CONTROL;
+		p->fr.subclass = AST_CONTROL_RING;
+		if (p->owner)
+			p->owner->state = AST_STATE_UP;
+		if (aopen_startrec(p))
+			return 	NULL;
+		return &p->fr;
+	case 'b': /* Busy signal */
+		p->fr.frametype = AST_FRAME_CONTROL;
+		p->fr.subclass = AST_CONTROL_BUSY;
+		return &p->fr;
+	case 'o': /* Overrun */
+		ast_log(LOG_WARNING, "Overflow on modem, flushing buffers\n");
+		if (ast_modem_send(p, "\0x10E", 2)) 
+			ast_log(LOG_WARNING, "Unable to flush buffers\n");
+		return &p->fr;	
+	case 'u': /* Underrun */
+		ast_log(LOG_WARNING, "Data underrun\n");
+		/* Fall Through */
+	case CHAR_ETX: /* End Transmission */
+	case 'd': /* Dialtone */
+	case 'c': /* Calling Tone */
+	case 'e': /* European version */
+	case 'a': /* Answer Tone */
+	case 'f': /* Bell Answer Tone */
+	case 'T': /* Timing mark */
+	case 't': /* Handset off hook */
+	case 'h': /* Handset hungup */
+	case 0: /* Pseudo signal */
+		/* Ignore */
+		return &p->fr;	
+	default:
+		ast_log(LOG_DEBUG, "Unknown Escaped character '%c' (%d)\n", esc, esc);
+	}
+	return &p->fr;
+}
+
+static struct ast_frame *aopen_read(struct ast_modem_pvt *p)
+{
+	char result[256];
+	short *b;
+	struct ast_frame *f=NULL;
+	int res;
+	int x;
+	if (p->ministate == STATE_COMMAND) {
+		/* Read the first two bytes, first, in case it's a control message */
+		fread(result, 1, 2, p->f);
+		if (result[0] == CHAR_DLE) {
+			return aopen_handle_escape(p, result[1]);
+			
+		} else {
+			if ((result[0] == '\n') || (result[0] == '\r'))
+				return aopen_handle_escape(p, 0);
+			/* Read the rest of the line */
+			fgets(result + 2, sizeof(result) - 2, p->f);
+			ast_modem_trim(result);
+			if (!strcasecmp(result, "VCON")) {
+				/* If we're in immediate mode, reply now */
+				if (p->mode == MODEM_MODE_IMMEDIATE)
+					return aopen_handle_escape(p, 'X');
+			} else
+			if (!strcasecmp(result, "BUSY")) {
+				/* Same as a busy signal */
+				return aopen_handle_escape(p, 'b');
+			} else
+			if (!strcasecmp(result, "RING")) {
+				return aopen_handle_escape(p, 'R');
+			} else
+			if (!strcasecmp(result, "NO DIALTONE")) {
+				/* There's no dialtone, so the line isn't working */
+				ast_log(LOG_WARNING, "Device '%s' lacking dialtone\n", p->dev);
+				return NULL;
+			}
+			ast_log(LOG_DEBUG, "Modem said '%s'\n", result);
+			return aopen_handle_escape(p, 0);
+		}
+	} else {
+		/* We have to be more efficient in voice mode */
+		b = (short *)(p->obuf + p->obuflen);
+		while (p->obuflen/2 < 240) {
+			/* Read ahead the full amount */
+			res = fread(result, 1, 240 - p->obuflen/2, p->f);
+			if (res < 1) {
+				/* If there's nothing there, just continue on */
+				if (errno == EAGAIN)
+					return aopen_handle_escape(p, 0);
+				ast_log(LOG_WARNING, "Read failed: %s\n", strerror(errno));
+			}
+			for (x=0;x<res;x++) {
+				/* Process all the bytes that we've read */
+				if (result[x] == CHAR_DLE) {
+					/* We assume there is no more than one signal frame among our
+					   data.  */
+					if (f) 
+						ast_log(LOG_WARNING, "Warning: Dropped a signal frame\n");
+					f = aopen_handle_escape(p, result[x+1]);
+					/* If aopen_handle_escape says NULL, say it now, doesn't matter
+					   what else is there, the connection is dead. */
+					if (!f)
+						return NULL;
+				} else {
+				/* Generate a 16-bit signed linear value from our 
+				   unsigned 8-bit value */
+					*(b++) = (((short)result[x]) - 127) * 0xff;
+					p->obuflen += 2;
+				}
+			}
+			if (f)
+				break;
+		}
+		/* If we have a control frame, return it now */
+		if (f)
+			return f;
+		/* If we get here, we have a complete voice frame */
+		p->fr.frametype = AST_FRAME_VOICE;
+		p->fr.subclass = AST_FORMAT_SLINEAR;
+		p->fr.timelen = 30;
+		p->fr.data = p->obuf;
+		p->fr.datalen = p->obuflen;
+		p->fr.mallocd = 0;
+		p->fr.offset = AST_FRIENDLY_OFFSET;
+		p->fr.src = __FUNCTION__;
+		if (option_debug)
+			ast_log(LOG_DEBUG, "aopen_read(voice frame)\n");
+		p->obuflen = 0;
+		return &p->fr;
+	}
+	return NULL;
+}
+
+static int aopen_write(struct ast_modem_pvt *p, struct ast_frame *f)
+{
+	if (option_debug)
+		ast_log(LOG_DEBUG, "aopen_write()\n");
+	return 0;
+}
+
+static char *aopen_identify(struct ast_modem_pvt *p)
+{
+	char identity[256];
+	char mfr[80];
+	char mdl[80];
+	char rev[80];
+	ast_modem_send(p, "AT#MDL?", 0);
+	ast_modem_read_response(p, 5);
+	strncpy(mdl, p->response, sizeof(mdl));
+	ast_modem_trim(mdl);
+	ast_modem_expect(p, "OK", 5);
+	ast_modem_send(p, "AT#MFR?", 0);
+	ast_modem_read_response(p, 5);
+	strncpy(mfr, p->response, sizeof(mfr));
+	ast_modem_trim(mfr);
+	ast_modem_expect(p, "OK", 5);
+	ast_modem_send(p, "AT#REV?", 0);
+	ast_modem_read_response(p, 5);
+	strncpy(rev, p->response, sizeof(rev));
+	ast_modem_trim(rev);
+	ast_modem_expect(p, "OK", 5);
+	snprintf(identity, sizeof(identity), "%s Model %s Revision %s", mfr, mdl, rev);
+	return strdup(identity);
+}
+
+static void aopen_incusecnt()
+{
+	pthread_mutex_lock(&usecnt_lock);
+	usecnt++;
+	pthread_mutex_unlock(&usecnt_lock);
+	ast_update_use_count();
+}
+
+static void aopen_decusecnt()
+{
+	pthread_mutex_lock(&usecnt_lock);
+	usecnt++;
+	pthread_mutex_unlock(&usecnt_lock);
+	ast_update_use_count();
+}
+
+static int aopen_answer(struct ast_modem_pvt *p)
+{
+	if (ast_modem_send(p, "ATA", 0) ||
+	     ast_modem_expect(p, "VCON", 10)) {
+		ast_log(LOG_WARNING, "Unable to answer: %s", p->response);
+		return -1;
+	}
+	return 0;
+}
+
+static int aopen_dialdigit(struct ast_modem_pvt *p, char digit)
+{
+	char cmd[80];
+	snprintf(cmd, sizeof(cmd), "AT#VTS=%c", digit);
+	if (ast_modem_send(p, cmd, 0) ||
+	     ast_modem_expect(p, "VCON", 10)) {
+		ast_log(LOG_WARNING, "Unable to answer: %s", p->response);
+		return -1;
+	}
+	return 0;
+}
+
+static int aopen_dial(struct ast_modem_pvt *p, char *stuff)
+{
+	char cmd[80];
+	snprintf(cmd, sizeof(cmd), "ATD%c %s", p->dialtype,stuff);
+	if (ast_modem_send(p, cmd, 0)) {
+		ast_log(LOG_WARNING, "Unable to dial\n");
+		return -1;
+	}
+	return 0;
+}
+
+static int aopen_hangup(struct ast_modem_pvt *p)
+{
+	if (aopen_break(p))
+		return -1;
+	/* Hangup by switching to data, then back to voice */
+	if (ast_modem_send(p, "ATH", 0) ||
+	     ast_modem_expect(p, "OK", 8)) {
+		ast_log(LOG_WARNING, "Unable to set to data mode\n");
+		return -1;
+	}
+	if (ast_modem_send(p, "AT#CLS=8", 0) ||
+	     ast_modem_expect(p, "OK", 5)) {
+		ast_log(LOG_WARNING, "Unable to set to voice mode\n");
+		return -1;
+	}
+	return 0;
+}
+
+static struct ast_modem_driver aopen_driver =
+{
+	"AOpen",
+	aopen_idents,
+	AST_FORMAT_SLINEAR,
+	0,		/* Not full duplex */
+	aopen_incusecnt,	/* incusecnt */
+	aopen_decusecnt,	/* decusecnt */
+	aopen_identify,	/* identify */
+	aopen_init,	/* init */
+	aopen_setdev,	/* setdev */
+	aopen_read,
+	aopen_write,
+	aopen_dial,	/* dial */
+	aopen_answer,	/* answer */
+	aopen_hangup,	/* hangup */
+	aopen_startrec,	/* start record */
+	NULL,	/* stop record */
+	NULL,	/* start playback */
+	NULL,	/* stop playback */
+	NULL,	/* set silence supression */
+	aopen_dialdigit,	/* dialdigit */
+};
+
+
+
+int usecount(void)
+{
+	int res;
+	pthread_mutex_lock(&usecnt_lock);
+	res = usecnt;
+	pthread_mutex_unlock(&usecnt_lock);
+	return res;
+}
+
+int load_module(void)
+{
+	return ast_register_modem_driver(&aopen_driver);
+}
+
+int unload_module(void)
+{
+	return ast_unregister_modem_driver(&aopen_driver);
+}
+
+char *description()
+{
+	return desc;
+}
+

channels/chan_oss.c

@@ -0,0 +1,843 @@
+/*
+ * Asterisk -- A telephony toolkit for Linux.
+ *
+ * Use /dev/dsp as a channel, and the console to command it :).
+ *
+ * The full-duplex "simulation" is pretty weak.  This is generally a 
+ * VERY BADLY WRITTEN DRIVER so please don't use it as a model for
+ * writing a driver.
+ * 
+ * Copyright (C) 1999, Mark Spencer
+ *
+ * Mark Spencer <markster@linux-support.net>
+ *
+ * This program is free software, distributed under the terms of
+ * the GNU General Public License
+ */
+
+#include <asterisk/frame.h>
+#include <asterisk/logger.h>
+#include <asterisk/channel.h>
+#include <asterisk/module.h>
+#include <asterisk/channel_pvt.h>
+#include <asterisk/options.h>
+#include <asterisk/pbx.h>
+#include <asterisk/config.h>
+#include <asterisk/cli.h>
+#include <unistd.h>
+#include <fcntl.h>
+#include <errno.h>
+#include <sys/ioctl.h>
+#include <sys/time.h>
+#include <string.h>
+#include <stdlib.h>
+#include <stdio.h>
+#include <linux/soundcard.h>
+
+/* Which device to use */
+#define DEV_DSP "/dev/dsp"
+
+/* Lets use 160 sample frames, just like GSM.  */
+#define FRAME_SIZE 160
+
+/* When you set the frame size, you have to come up with
+   the right buffer format as well. */
+/* 5 64-byte frames = one frame */
+#define BUFFER_FMT ((buffersize * 5) << 16) | (0x0006);
+
+/* Don't switch between read/write modes faster than every 300 ms */
+#define MIN_SWITCH_TIME 600
+
+static struct timeval lasttime;
+
+static int usecnt;
+static int needanswer = 0;
+static int needhangup = 0;
+static int silencesuppression = 0;
+static int silencethreshold = 1000;
+
+static char digits[80] = "";
+static char text2send[80] = "";
+
+static pthread_mutex_t usecnt_lock = PTHREAD_MUTEX_INITIALIZER;
+
+static char *type = "Console";
+static char *desc = "OSS Console Channel Driver";
+static char *tdesc = "OSS Console Channel Driver";
+static char *config = "oss.conf";
+
+static char context[AST_MAX_EXTENSION] = "default";
+static char language[MAX_LANGUAGE] = "";
+static char exten[AST_MAX_EXTENSION] = "s";
+
+/* Some pipes to prevent overflow */
+static int funnel[2];
+static pthread_mutex_t sound_lock = PTHREAD_MUTEX_INITIALIZER;
+static pthread_t silly;
+
+static struct chan_oss_pvt {
+	/* We only have one OSS structure -- near sighted perhaps, but it
+	   keeps this driver as simple as possible -- as it should be. */
+	struct ast_channel *owner;
+	char exten[AST_MAX_EXTENSION];
+	char context[AST_MAX_EXTENSION];
+} oss;
+
+static int time_has_passed()
+{
+	struct timeval tv;
+	int ms;
+	gettimeofday(&tv, NULL);
+	ms = (tv.tv_sec - lasttime.tv_sec) * 1000 +
+			(tv.tv_usec - lasttime.tv_usec) / 1000;
+	if (ms > MIN_SWITCH_TIME)
+		return -1;
+	return 0;
+}
+
+/* Number of buffers...  Each is FRAMESIZE/8 ms long.  For example
+   with 160 sample frames, and a buffer size of 3, we have a 60ms buffer, 
+   usually plenty. */
+
+
+#define MAX_BUFFER_SIZE 100
+static int buffersize = 3;
+
+static int full_duplex = 0;
+
+/* Are we reading or writing (simulated full duplex) */
+static int readmode = 1;
+
+/* File descriptor for sound device */
+static int sounddev = -1;
+
+static int autoanswer = 1;
+ 
+static int calc_loudness(short *frame)
+{
+	int sum = 0;
+	int x;
+	for (x=0;x<FRAME_SIZE;x++) {
+		if (frame[x] < 0)
+			sum -= frame[x];
+		else
+			sum += frame[x];
+	}
+	sum = sum/FRAME_SIZE;
+	return sum;
+}
+
+static int silence_suppress(short *buf)
+{
+#define SILBUF 3
+	int loudness;
+	static int silentframes = 0;
+	static char silbuf[FRAME_SIZE * 2 * SILBUF];
+	static int silbufcnt=0;
+	if (!silencesuppression)
+		return 0;
+	loudness = calc_loudness((short *)(buf));
+	if (option_debug)
+		ast_log(LOG_DEBUG, "loudness is %d\n", loudness);
+	if (loudness < silencethreshold) {
+		silentframes++;
+		silbufcnt++;
+		/* Keep track of the last few bits of silence so we can play
+		   them as lead-in when the time is right */
+		if (silbufcnt >= SILBUF) {
+			/* Make way for more buffer */
+			memmove(silbuf, silbuf + FRAME_SIZE * 2, FRAME_SIZE * 2 * (SILBUF - 1));
+			silbufcnt--;
+		}
+		memcpy(silbuf + FRAME_SIZE * 2 * silbufcnt, buf, FRAME_SIZE * 2);
+		if (silentframes > 10) {
+			/* We've had plenty of silence, so compress it now */
+			return 1;
+		}
+	} else {
+		silentframes=0;
+		/* Write any buffered silence we have, it may have something
+		   important */
+		if (silbufcnt) {
+			write(funnel[1], silbuf, silbufcnt * FRAME_SIZE);
+			silbufcnt = 0;
+		}
+	}
+	return 0;
+}
+
+static void *silly_thread(void *ignore)
+{
+	char buf[FRAME_SIZE * 2];
+	int pos=0;
+	int res=0;
+	/* Read from the sound device, and write to the pipe. */
+	for (;;) {
+		/* Give the writer a better shot at the lock */
+#if 0
+		usleep(1000);
+#endif		
+		pthread_testcancel();
+		pthread_mutex_lock(&sound_lock);
+		res = read(sounddev, buf + pos, FRAME_SIZE * 2 - pos);
+		pthread_mutex_unlock(&sound_lock);
+		if (res > 0) {
+			pos += res;
+			if (pos == FRAME_SIZE * 2) {
+				if (needhangup || needanswer || strlen(digits) || 
+				    !silence_suppress((short *)buf)) {
+					res = write(funnel[1], buf, sizeof(buf));
+				}
+				pos = 0;
+			}
+		} else {
+			close(funnel[1]);
+			break;
+		}
+		pthread_testcancel();
+	}
+	return NULL;
+}
+
+static int setformat(void)
+{
+	int fmt, desired, res, fd = sounddev;
+	static int warnedalready = 0;
+	static int warnedalready2 = 0;
+	pthread_mutex_lock(&sound_lock);
+	fmt = AFMT_S16_LE;
+	res = ioctl(fd, SNDCTL_DSP_SETFMT, &fmt);
+	if (res < 0) {
+		ast_log(LOG_WARNING, "Unable to set format to 16-bit signed\n");
+		pthread_mutex_unlock(&sound_lock);
+		return -1;
+	}
+	res = ioctl(fd, SNDCTL_DSP_SETDUPLEX, 0);
+	if (res >= 0) {
+		if (option_verbose > 1) 
+			ast_verbose(VERBOSE_PREFIX_2 "Console is full duplex\n");
+		full_duplex = -1;
+	}
+	fmt = 0;
+	res = ioctl(fd, SNDCTL_DSP_STEREO, &fmt);
+	if (res < 0) {
+		ast_log(LOG_WARNING, "Failed to set audio device to mono\n");
+		pthread_mutex_unlock(&sound_lock);
+		return -1;
+	}
+	/* 8000 Hz desired */
+	desired = 8000;
+	fmt = desired;
+	res = ioctl(fd, SNDCTL_DSP_SPEED, &fmt);
+	if (res < 0) {
+		ast_log(LOG_WARNING, "Failed to set audio device to mono\n");
+		pthread_mutex_unlock(&sound_lock);
+		return -1;
+	}
+	if (fmt != desired) {
+		if (!warnedalready++)
+			ast_log(LOG_WARNING, "Requested %d Hz, got %d Hz -- sound may be choppy\n", desired, fmt);
+	}
+#if 1
+	fmt = BUFFER_FMT;
+	res = ioctl(fd, SNDCTL_DSP_SETFRAGMENT, &fmt);
+	if (res < 0) {
+		if (!warnedalready2++)
+			ast_log(LOG_WARNING, "Unable to set fragment size -- sound may be choppy\n");
+	}
+#endif
+	pthread_mutex_unlock(&sound_lock);
+	return 0;
+}
+
+static int soundcard_setoutput(int force)
+{
+	/* Make sure the soundcard is in output mode.  */
+	int fd = sounddev;
+	if (full_duplex || (!readmode && !force))
+		return 0;
+	pthread_mutex_lock(&sound_lock);
+	readmode = 0;
+	if (force || time_has_passed()) {
+		ioctl(sounddev, SNDCTL_DSP_RESET);
+		/* Keep the same fd reserved by closing the sound device and copying stdin at the same
+		   time. */
+		/* dup2(0, sound); */ 
+		close(sounddev);
+		fd = open(DEV_DSP, O_WRONLY);
+		if (fd < 0) {
+			ast_log(LOG_WARNING, "Unable to re-open DSP device: %s\n", strerror(errno));
+			pthread_mutex_unlock(&sound_lock);
+			return -1;
+		}
+		/* dup2 will close the original and make fd be sound */
+		if (dup2(fd, sounddev) < 0) {
+			ast_log(LOG_WARNING, "dup2() failed: %s\n", strerror(errno));
+			pthread_mutex_unlock(&sound_lock);
+			return -1;
+		}
+		if (setformat()) {
+			pthread_mutex_unlock(&sound_lock);
+			return -1;
+		}
+		pthread_mutex_unlock(&sound_lock);
+		return 0;
+	}
+	pthread_mutex_unlock(&sound_lock);
+	return 1;
+}
+
+static int soundcard_setinput(int force)
+{
+	int fd = sounddev;
+	if (full_duplex || (readmode && !force))
+		return 0;
+	pthread_mutex_lock(&sound_lock);
+	readmode = -1;
+	if (force || time_has_passed()) {
+		ioctl(sounddev, SNDCTL_DSP_RESET);
+		close(sounddev);
+		/* dup2(0, sound); */
+		fd = open(DEV_DSP, O_RDONLY);
+		if (fd < 0) {
+			ast_log(LOG_WARNING, "Unable to re-open DSP device: %s\n", strerror(errno));
+			pthread_mutex_unlock(&sound_lock);
+			return -1;
+		}
+		/* dup2 will close the original and make fd be sound */
+		if (dup2(fd, sounddev) < 0) {
+			ast_log(LOG_WARNING, "dup2() failed: %s\n", strerror(errno));
+			pthread_mutex_unlock(&sound_lock);
+			return -1;
+		}
+		if (setformat()) {
+			pthread_mutex_unlock(&sound_lock);
+			return -1;
+		}
+		pthread_mutex_unlock(&sound_lock);
+		return 0;
+	}
+	pthread_mutex_unlock(&sound_lock);
+	return 1;
+}
+
+static int soundcard_init()
+{
+	/* Assume it's full duplex for starters */
+	int fd = open(DEV_DSP, 	O_RDWR);
+	if (fd < 0) {
+		ast_log(LOG_ERROR, "Unable to open %s: %s\n", DEV_DSP, strerror(errno));
+		return fd;
+	}
+	gettimeofday(&lasttime, NULL);
+	sounddev = fd;
+	setformat();
+	if (!full_duplex) 
+		soundcard_setinput(1);
+	return sounddev;
+}
+
+static int oss_digit(struct ast_channel *c, char digit)
+{
+	ast_verbose( " << Console Received digit %c >> \n", digit);
+	return 0;
+}
+
+static int oss_text(struct ast_channel *c, char *text)
+{
+	ast_verbose( " << Console Received text %s >> \n", text);
+	return 0;
+}
+
+static int oss_call(struct ast_channel *c, char *dest, int timeout)
+{
+	ast_verbose( " << Call placed to '%s' on console >> \n", dest);
+	if (autoanswer) {
+		ast_verbose( " << Auto-answered >> \n" );
+		needanswer = 1;
+	} else {
+		ast_verbose( " << Type 'answer' to answer, or use 'autoanswer' for future calls >> \n");
+	}
+	return 0;
+}
+
+static int oss_answer(struct ast_channel *c)
+{
+	ast_verbose( " << Console call has been answered >> \n");
+	c->state = AST_STATE_UP;
+	return 0;
+}
+
+static int oss_hangup(struct ast_channel *c)
+{
+	c->pvt->pvt = NULL;
+	oss.owner = NULL;
+	ast_verbose( " << Hangup on console >> \n");
+	pthread_mutex_lock(&usecnt_lock);
+	usecnt--;
+	pthread_mutex_unlock(&usecnt_lock);
+	needhangup = 0;
+	needanswer = 0;
+	return 0;
+}
+
+static int soundcard_writeframe(short *data)
+{	
+	/* Write an exactly FRAME_SIZE sized of frame */
+	static int bufcnt = 0;
+	static char buffer[FRAME_SIZE * 2 * MAX_BUFFER_SIZE * 5];
+	struct audio_buf_info info;
+	int res;
+	int fd = sounddev;
+	static int warned=0;
+	pthread_mutex_lock(&sound_lock);
+	if (ioctl(fd, SNDCTL_DSP_GETOSPACE, &info)) {
+		if (!warned)
+			ast_log(LOG_WARNING, "Error reading output space\n");
+		bufcnt = buffersize;
+		warned++;
+	}
+	if ((info.fragments >= buffersize * 5) && (bufcnt == buffersize)) {
+		/* We've run out of stuff, buffer again */
+		bufcnt = 0;
+	}
+	if (bufcnt == buffersize) {
+		/* Write sample immediately */
+		res = write(fd, ((void *)data), FRAME_SIZE * 2);
+	} else {
+		/* Copy the data into our buffer */
+		res = FRAME_SIZE * 2;
+		memcpy(buffer + (bufcnt * FRAME_SIZE * 2), data, FRAME_SIZE * 2);
+		bufcnt++;
+		if (bufcnt == buffersize) {
+			res = write(fd, ((void *)buffer), FRAME_SIZE * 2 * buffersize);
+		}
+	}
+	pthread_mutex_unlock(&sound_lock);
+	return res;
+}
+
+
+static int oss_write(struct ast_channel *chan, struct ast_frame *f)
+{
+	int res;
+	static char sizbuf[8000];
+	static int sizpos = 0;
+	int len = sizpos;
+	int pos;
+	if (!full_duplex && (strlen(digits) || needhangup || needanswer)) {
+		/* If we're half duplex, we have to switch to read mode
+		   to honor immediate needs if necessary */
+		res = soundcard_setinput(1);
+		if (res < 0) {
+			ast_log(LOG_WARNING, "Unable to set device to input mode\n");
+			return -1;
+		}
+		return 0;
+	}
+	res = soundcard_setoutput(0);
+	if (res < 0) {
+		ast_log(LOG_WARNING, "Unable to set output device\n");
+		return -1;
+	} else if (res > 0) {
+		/* The device is still in read mode, and it's too soon to change it,
+		   so just pretend we wrote it */
+		return 0;
+	}
+	/* We have to digest the frame in 160-byte portions */
+	if (f->datalen > sizeof(sizbuf) - sizpos) {
+		ast_log(LOG_WARNING, "Frame too large\n");
+		return -1;
+	}
+	memcpy(sizbuf + sizpos, f->data, f->datalen);
+	len += f->datalen;
+	pos = 0;
+	while(len - pos > FRAME_SIZE * 2) {
+		soundcard_writeframe((short *)(sizbuf + pos));
+		pos += FRAME_SIZE * 2;
+	}
+	if (len - pos) 
+		memmove(sizbuf, sizbuf + pos, len - pos);
+	sizpos = len - pos;
+	return 0;
+}
+
+static struct ast_frame *oss_read(struct ast_channel *chan)
+{
+	static struct ast_frame f;
+	static char buf[FRAME_SIZE * 2 + AST_FRIENDLY_OFFSET];
+	static int readpos = 0;
+	int res;
+	
+#if 0
+	ast_log(LOG_DEBUG, "oss_read()\n");
+#endif
+	
+	f.frametype = AST_FRAME_NULL;
+	f.subclass = 0;
+	f.timelen = 0;
+	f.datalen = 0;
+	f.data = NULL;
+	f.offset = 0;
+	f.src = type;
+	f.mallocd = 0;
+	
+	if (needhangup) {
+		return NULL;
+	}
+	if (strlen(text2send)) {
+		f.frametype = AST_FRAME_TEXT;
+		f.subclass = 0;
+		f.data = text2send;
+		f.datalen = strlen(text2send);
+		strcpy(text2send,"");
+		return &f;
+	}
+	if (strlen(digits)) {
+		f.frametype = AST_FRAME_DTMF;
+		f.subclass = digits[0];
+		for (res=0;res<strlen(digits);res++)
+			digits[res] = digits[res + 1];
+		return &f;
+	}
+	
+	if (needanswer) {
+		needanswer = 0;
+		f.frametype = AST_FRAME_CONTROL;
+		f.subclass = AST_CONTROL_ANSWER;
+		chan->state = AST_STATE_UP;
+		return &f;
+	}
+	
+	res = soundcard_setinput(0);
+	if (res < 0) {
+		ast_log(LOG_WARNING, "Unable to set input mode\n");
+		return NULL;
+	}
+	if (res > 0) {
+		/* Theoretically shouldn't happen, but anyway, return a NULL frame */
+		return &f;
+	}
+	res = read(funnel[0], buf + AST_FRIENDLY_OFFSET + readpos, FRAME_SIZE * 2 - readpos);
+	if (res < 0) {
+		ast_log(LOG_WARNING, "Error reading from sound device: %s\n", strerror(errno));
+		return NULL;
+	}
+	readpos += res;
+	
+	if (readpos == FRAME_SIZE * 2) {
+		/* A real frame */
+		readpos = 0;
+		f.frametype = AST_FRAME_VOICE;
+		f.subclass = AST_FORMAT_SLINEAR;
+		f.timelen = FRAME_SIZE / 8;
+		f.datalen = FRAME_SIZE * 2;
+		f.data = buf + AST_FRIENDLY_OFFSET;
+		f.offset = AST_FRIENDLY_OFFSET;
+		f.src = type;
+		f.mallocd = 0;
+	}
+	return &f;
+}
+
+static struct ast_channel *oss_new(struct chan_oss_pvt *p, int state)
+{
+	struct ast_channel *tmp;
+	tmp = ast_channel_alloc();
+	if (tmp) {
+		snprintf(tmp->name, sizeof(tmp->name), "OSS/%s", DEV_DSP + 5);
+		tmp->type = type;
+		tmp->fd = funnel[0];
+		tmp->format = AST_FORMAT_SLINEAR;
+		tmp->pvt->pvt = p;
+		tmp->pvt->send_digit = oss_digit;
+		tmp->pvt->send_text = oss_text;
+		tmp->pvt->hangup = oss_hangup;
+		tmp->pvt->answer = oss_answer;
+		tmp->pvt->read = oss_read;
+		tmp->pvt->call = oss_call;
+		tmp->pvt->write = oss_write;
+		if (strlen(p->context))
+			strncpy(tmp->context, p->context, sizeof(tmp->context));
+		if (strlen(p->exten))
+			strncpy(tmp->exten, p->exten, sizeof(tmp->exten));
+		if (strlen(language))
+			strncpy(tmp->language, language, sizeof(tmp->language));
+		p->owner = tmp;
+		tmp->state = state;
+		pthread_mutex_lock(&usecnt_lock);
+		usecnt++;
+		pthread_mutex_unlock(&usecnt_lock);
+		ast_update_use_count();
+		if (state != AST_STATE_DOWN) {
+			if (ast_pbx_start(tmp)) {
+				ast_log(LOG_WARNING, "Unable to start PBX on %s\n", tmp->name);
+				ast_hangup(tmp);
+				tmp = NULL;
+			}
+		}
+	}
+	return tmp;
+}
+
+static struct ast_channel *oss_request(char *type, int format, void *data)
+{
+	int oldformat = format;
+	struct ast_channel *tmp;
+	format &= AST_FORMAT_SLINEAR;
+	if (!format) {
+		ast_log(LOG_NOTICE, "Asked to get a channel of format '%d'\n", oldformat);
+		return NULL;
+	}
+	if (oss.owner) {
+		ast_log(LOG_NOTICE, "Already have a call on the OSS channel\n");
+		return NULL;
+	}
+	tmp= oss_new(&oss, AST_STATE_DOWN);
+	if (!tmp) {
+		ast_log(LOG_WARNING, "Unable to create new OSS channel\n");
+	}
+	return tmp;
+}
+
+static int console_autoanswer(int fd, int argc, char *argv[])
+{
+	if ((argc != 1) && (argc != 2))
+		return RESULT_SHOWUSAGE;
+	if (argc == 1) {
+		ast_cli(fd, "Auto answer is %s.\n", autoanswer ? "on" : "off");
+		return RESULT_SUCCESS;
+	} else {
+		if (!strcasecmp(argv[1], "on"))
+			autoanswer = -1;
+		else if (!strcasecmp(argv[1], "off"))
+			autoanswer = 0;
+		else
+			return RESULT_SHOWUSAGE;
+	}
+	return RESULT_SUCCESS;
+}
+
+static char *autoanswer_complete(char *line, char *word, int pos, int state)
+{
+#ifndef MIN
+#define MIN(a,b) ((a) < (b) ? (a) : (b))
+#endif
+	switch(state) {
+	case 0:
+		if (strlen(word) && !strncasecmp(word, "on", MIN(strlen(word), 2)))
+			return strdup("on");
+	case 1:
+		if (strlen(word) && !strncasecmp(word, "off", MIN(strlen(word), 3)))
+			return strdup("off");
+	default:
+		return NULL;
+	}
+	return NULL;
+}
+
+static char autoanswer_usage[] =
+"Usage: autoanswer [on|off]\n"
+"       Enables or disables autoanswer feature.  If used without\n"
+"       argument, displays the current on/off status of autoanswer.\n"
+"       The default value of autoanswer is in 'oss.conf'.\n";
+
+static int console_answer(int fd, int argc, char *argv[])
+{
+	if (argc != 1)
+		return RESULT_SHOWUSAGE;
+	if (!oss.owner) {
+		ast_cli(fd, "No one is calling us\n");
+		return RESULT_FAILURE;
+	}
+	needanswer++;
+	return RESULT_SUCCESS;
+}
+
+static char sendtext_usage[] =
+"Usage: send text <message>\n"
+"       Sends a text message for display on the remote terminal.\n";
+
+static int console_sendtext(int fd, int argc, char *argv[])
+{
+	int tmparg = 1;
+	if (argc < 1)
+		return RESULT_SHOWUSAGE;
+	if (!oss.owner) {
+		ast_cli(fd, "No one is calling us\n");
+		return RESULT_FAILURE;
+	}
+	if (strlen(text2send))
+		ast_cli(fd, "Warning: message already waiting to be sent, overwriting\n");
+	strcpy(text2send, "");
+	while(tmparg <= argc) {
+		strncat(text2send, argv[tmparg++], sizeof(text2send) - strlen(text2send));
+		strncat(text2send, " ", sizeof(text2send) - strlen(text2send));
+	}
+	needanswer++;
+	return RESULT_SUCCESS;
+}
+
+static char answer_usage[] =
+"Usage: answer\n"
+"       Answers an incoming call on the console (OSS) channel.\n";
+
+static int console_hangup(int fd, int argc, char *argv[])
+{
+	if (argc != 1)
+		return RESULT_SHOWUSAGE;
+	if (!oss.owner) {
+		ast_cli(fd, "No call to hangup up\n");
+		return RESULT_FAILURE;
+	}
+	needhangup++;
+	return RESULT_SUCCESS;
+}
+
+static char hangup_usage[] =
+"Usage: hangup\n"
+"       Hangs up any call currently placed on the console.\n";
+
+
+static int console_dial(int fd, int argc, char *argv[])
+{
+	char tmp[256], *tmp2;
+	char *mye, *myc;
+	if ((argc != 1) && (argc != 2))
+		return RESULT_SHOWUSAGE;
+	if (oss.owner) {
+		if (argc == 2)
+			strncat(digits, argv[1], sizeof(digits) - strlen(digits));
+		else {
+			ast_cli(fd, "You're already in a call.  You can use this only to dial digits until you hangup\n");
+			return RESULT_FAILURE;
+		}
+		return RESULT_SUCCESS;
+	}
+	mye = exten;
+	myc = context;
+	if (argc == 2) {
+		strncpy(tmp, argv[1], sizeof(tmp));
+		strtok(tmp, "@");
+		tmp2 = strtok(NULL, "@");
+		if (strlen(tmp))
+			mye = tmp;
+		if (tmp2 && strlen(tmp2))
+			myc = tmp2;
+	}
+	if (ast_exists_extension(NULL, myc, mye, 1)) {
+		strncpy(oss.exten, mye, sizeof(oss.exten));
+		strncpy(oss.context, myc, sizeof(oss.context));
+		oss_new(&oss, AST_STATE_UP);
+	} else
+		ast_cli(fd, "No such extension '%s' in context '%s'\n", mye, myc);
+	return RESULT_SUCCESS;
+}
+
+static char dial_usage[] =
+"Usage: dial [extension[@context]]\n"
+"       Dials a given extensison (";
+
+
+static struct ast_cli_entry myclis[] = {
+	{ { "answer", NULL }, console_answer, "Answer an incoming console call", answer_usage },
+	{ { "hangup", NULL }, console_hangup, "Hangup a call on the console", hangup_usage },
+	{ { "dial", NULL }, console_dial, "Dial an extension on the console", dial_usage },
+	{ { "send text", NULL }, console_sendtext, "Send text to the remote device", sendtext_usage },
+	{ { "autoanswer", NULL }, console_autoanswer, "Sets/displays autoanswer", autoanswer_usage, autoanswer_complete }
+};
+
+int load_module()
+{
+	int res;
+	int x;
+	int flags;
+	struct ast_config *cfg = ast_load(config);
+	struct ast_variable *v;
+	res = pipe(funnel);
+	if (res) {
+		ast_log(LOG_ERROR, "Unable to create pipe\n");
+		return -1;
+	}
+	/* We make the funnel so that writes to the funnel don't block...
+	   Our "silly" thread can read to its heart content, preventing
+	   recording overruns */
+	flags = fcntl(funnel[1], F_GETFL);
+#if 0
+	fcntl(funnel[0], F_SETFL, flags | O_NONBLOCK);
+#endif
+	fcntl(funnel[1], F_SETFL, flags | O_NONBLOCK);
+	res = soundcard_init();
+	if (res < 0) {
+		close(funnel[1]);
+		close(funnel[0]);
+		return -1;
+	}
+	if (!full_duplex)
+		ast_log(LOG_WARNING, "XXX I don't work right with non-full duplex sound cards XXX\n");
+	pthread_create(&silly, NULL, silly_thread, NULL);
+	res = ast_channel_register(type, tdesc, AST_FORMAT_SLINEAR, oss_request);
+	if (res < 0) {
+		ast_log(LOG_ERROR, "Unable to register channel class '%s'\n", type);
+		return -1;
+	}
+	for (x=0;x<sizeof(myclis)/sizeof(struct ast_cli_entry); x++)
+		ast_cli_register(myclis + x);
+	if (cfg) {
+		v = ast_variable_browse(cfg, "general");
+		while(v) {
+			if (!strcasecmp(v->name, "autoanswer"))
+				autoanswer = ast_true(v->value);
+			else if (!strcasecmp(v->name, "silencesuppression"))
+				silencesuppression = ast_true(v->value);
+			else if (!strcasecmp(v->name, "silencethreshold"))
+				silencethreshold = atoi(v->value);
+			else if (!strcasecmp(v->name, "context"))
+				strncpy(context, v->value, sizeof(context));
+			else if (!strcasecmp(v->name, "language"))
+				strncpy(language, v->value, sizeof(language));
+			else if (!strcasecmp(v->name, "extension"))
+				strncpy(exten, v->value, sizeof(exten));
+			v=v->next;
+		}
+		ast_destroy(cfg);
+	}
+	return 0;
+}
+
+
+
+int unload_module()
+{
+	int x;
+	for (x=0;x<sizeof(myclis)/sizeof(struct ast_cli_entry); x++)
+		ast_cli_unregister(myclis + x);
+	close(sounddev);
+	if (funnel[0] > 0) {
+		close(funnel[0]);
+		close(funnel[1]);
+	}
+	if (silly) {
+		pthread_cancel(silly);
+		pthread_join(silly, NULL);
+	}
+	if (oss.owner)
+		ast_softhangup(oss.owner);
+	if (oss.owner)
+		return -1;
+	return 0;
+}
+
+char *description()
+{
+	return desc;
+}
+
+int usecount()
+{
+	int res;
+	pthread_mutex_lock(&usecnt_lock);
+	res = usecnt;
+	pthread_mutex_unlock(&usecnt_lock);
+	return res;
+}

channels/chan_phone.c

@@ -1,9 +1,9 @@
 /*
  * Asterisk -- A telephony toolkit for Linux.
  *
- * QuickNet Internet Phone Jack Channel
+ * Generic Linux Telephony Interface driver
  * 
- * Copyright (C) 1999, Adtran Inc. and Linux Support Services, LLC
+ * Copyright (C) 1999, Mark Spencer
  *
  * Mark Spencer <markster@linux-support.net>
  *
@@ -29,27 +29,34 @@
 #include <arpa/inet.h>
 #include <fcntl.h>
 #include <sys/ioctl.h>
-#include <linux/if_packet.h>
-#include <linux/if_ether.h>
-#include "ixjuser.h"
+#include <linux/telephony.h>
+/* Still use some IXJ specific stuff */
+#include <linux/ixjuser.h>
 #include "DialTone.h"
 
-#define IXJ_MAX_BUF 480
+#define PHONE_MAX_BUF 480
 
-static char *desc = "QuickNet Internet Phone Jack";
-static char *type = "PhoneJack";
-static char *tdesc = "QuickNet Internet Phone Jack";
-static char *config = "ixj.conf";
+static char *desc = "Linux Telephony API Support";
+static char *type = "Phone";
+static char *tdesc = "Standard Linux Telephony API Driver";
+static char *config = "phone.conf";
 
 /* Default context for dialtone mode */
 static char context[AST_MAX_EXTENSION] = "default";
 
-char *ignore_rcs_id_for_chan_ixj = ixjuser_h_rcsid;
-
+/* Default language */
+static char language[MAX_LANGUAGE] = "";
 static int usecnt =0;
+
+static int echocancel = AEC_OFF;
+
+static int silencesupression = 0;
+
+static int prefformat = AST_FORMAT_G723_1 | AST_FORMAT_SLINEAR;
+
 static pthread_mutex_t usecnt_lock = PTHREAD_MUTEX_INITIALIZER;
 
-/* Protect the interface list (of ixj_pvt's) */
+/* Protect the interface list (of phone_pvt's) */
 static pthread_mutex_t iflock = PTHREAD_MUTEX_INITIALIZER;
 
 /* Protect the monitoring thread, so only one process can kill or start it, and not
@@ -60,15 +67,16 @@ static pthread_mutex_t monlock = PTHREAD_MUTEX_INITIALIZER;
    which are not currently in use.  */
 static pthread_t monitor_thread = -1;
 
-static int restart_monitor();
+static int restart_monitor(void);
 
 /* The private structures of the Phone Jack channels are linked for
    selecting outgoing channels */
    
 #define MODE_DIALTONE 	1
 #define MODE_IMMEDIATE	2
+#define MODE_FXO		3
    
-static struct ixj_pvt {
+static struct phone_pvt {
 	int fd;							/* Raw file descriptor for this device */
 	struct ast_channel *owner;		/* Channel we belong to, possibly NULL */
 	int mode;						/* Is this in the  */
@@ -76,20 +84,22 @@ static struct ixj_pvt {
 	int lastinput;					/* Last input format */
 	int ministate;					/* Miniature state, for dialtone mode */
 	char dev[256];					/* Device name */
-	struct ixj_pvt *next;			/* Next channel in list */
+	struct phone_pvt *next;			/* Next channel in list */
 	struct ast_frame fr;			/* Frame */
 	char offset[AST_FRIENDLY_OFFSET];
-	char buf[IXJ_MAX_BUF];					/* Static buffer for reading frames */
+	char buf[PHONE_MAX_BUF];					/* Static buffer for reading frames */
 	int obuflen;
 	int dialtone;
+	int silencesupression;
 	char context[AST_MAX_EXTENSION];
-	char obuf[IXJ_MAX_BUF * 2];
+	char obuf[PHONE_MAX_BUF * 2];
 	char ext[AST_MAX_EXTENSION];
+	char language[MAX_LANGUAGE];
 } *iflist = NULL;
 
-static int ixj_digit(struct ast_channel *ast, char digit)
+static int phone_digit(struct ast_channel *ast, char digit)
 {
-	struct ixj_pvt *p;
+	struct phone_pvt *p;
 	int outdigit;
 	p = ast->pvt->pvt;
 	switch(digit) {
@@ -115,55 +125,67 @@ static int ixj_digit(struct ast_channel *ast, char digit)
 		ast_log(LOG_WARNING, "Unknown digit '%c'\n", digit);
 		return -1;
 	}
-	ioctl(p->fd, IXJCTL_PLAY_TONE, digit);
+	ioctl(p->fd, PHONE_PLAY_TONE, digit);
 	return 0;
 }
 
-static int ixj_call(struct ast_channel *ast, char *dest, int timeout)
+static int phone_call(struct ast_channel *ast, char *dest, int timeout)
 {
-	struct ixj_pvt *p;
+	struct phone_pvt *p;
 	p = ast->pvt->pvt;
 	if ((ast->state != AST_STATE_DOWN) && (ast->state != AST_STATE_RESERVED)) {
-		ast_log(LOG_WARNING, "ixj_call called on %s, neither down nor reserved\n", ast->name);
+		ast_log(LOG_WARNING, "phone_call called on %s, neither down nor reserved\n", ast->name);
 		return -1;
 	}
 	/* When we call, it just works, really, there's no destination...  Just
 	   ring the phone and wait for someone to answer */
-	ast_log(LOG_DEBUG, "Ringing %s on %s (%d)\n", dest, ast->name, ast->fd);
-	ioctl(p->fd, IXJCTL_RING_START);
+	if (option_debug)
+		ast_log(LOG_DEBUG, "Ringing %s on %s (%d)\n", dest, ast->name, ast->fd);
+	ioctl(p->fd, PHONE_RING_START);
 	ast->state = AST_STATE_RINGING;
 	return 0;
 }
 
-static int ixj_hangup(struct ast_channel *ast)
+static int phone_hangup(struct ast_channel *ast)
 {
-	struct ixj_pvt *p;
+	struct phone_pvt *p;
 	p = ast->pvt->pvt;
-	ast_log(LOG_DEBUG, "ixj_hangup(%s)\n", ast->name);
+	if (option_debug)
+		ast_log(LOG_DEBUG, "phone_hangup(%s)\n", ast->name);
 	if (!ast->pvt->pvt) {
 		ast_log(LOG_WARNING, "Asked to hangup channel not connected\n");
 		return 0;
 	}
 	/* XXX Is there anything we can do to really hang up except stop recording? */
 	ast->state = AST_STATE_DOWN;
-	if (ioctl(p->fd, IXJCTL_REC_STOP))
+	if (ioctl(p->fd, PHONE_REC_STOP))
 		ast_log(LOG_WARNING, "Failed to stop recording\n");
-	if (ioctl(p->fd, IXJCTL_PLAY_STOP))
+	if (ioctl(p->fd, PHONE_PLAY_STOP))
 		ast_log(LOG_WARNING, "Failed to stop playing\n");
-	if (ioctl(p->fd, IXJCTL_RING_STOP))
+	if (ioctl(p->fd, PHONE_RING_STOP))
 		ast_log(LOG_WARNING, "Failed to stop ringing\n");
-	if (ioctl(p->fd, IXJCTL_CPT_STOP))
+	if (ioctl(p->fd, PHONE_CPT_STOP))
 		ast_log(LOG_WARNING, "Failed to stop sounds\n");
+
+	/* If it's an FXO, hang them up */
+	if (p->mode == MODE_FXO) {
+		if (ioctl(p->fd, PHONE_PSTN_SET_STATE, PSTN_ON_HOOK)) 
+			ast_log(LOG_DEBUG, "ioctl(PHONE_PSTN_SET_STATE) failed on %s (%s)\n",ast->name, strerror(errno));
+	}
+
 	/* If they're off hook, give a busy signal */
-	if (ioctl(p->fd, IXJCTL_HOOKSTATE))
-		ioctl(p->fd, IXJCTL_BUSY);
+	if (ioctl(p->fd, PHONE_HOOKSTATE)) {
+		if (option_debug)
+			ast_log(LOG_DEBUG, "Got hunghup, giving busy signal\n");
+		ioctl(p->fd, PHONE_BUSY);
+	}
 	p->lastformat = -1;
 	p->lastinput = -1;
 	p->ministate = 0;
 	p->obuflen = 0;
 	p->dialtone = 0;
 	memset(p->ext, 0, sizeof(p->ext));
-	((struct ixj_pvt *)(ast->pvt->pvt))->owner = NULL;
+	((struct phone_pvt *)(ast->pvt->pvt))->owner = NULL;
 	pthread_mutex_lock(&usecnt_lock);
 	usecnt--;
 	if (usecnt < 0) 
@@ -178,27 +200,27 @@ static int ixj_hangup(struct ast_channel *ast)
 	return 0;
 }
 
-static int ixj_setup(struct ast_channel *ast)
+static int phone_setup(struct ast_channel *ast)
 {
-	struct ixj_pvt *p;
+	struct phone_pvt *p;
 	p = ast->pvt->pvt;
-	ioctl(p->fd, IXJCTL_CPT_STOP);
+	ioctl(p->fd, PHONE_CPT_STOP);
 	/* Nothing to answering really, just start recording */
 	if (ast->format & AST_FORMAT_G723_1) {
 		/* Prefer g723 */
-		ioctl(p->fd, IXJCTL_REC_STOP);
+		ioctl(p->fd, PHONE_REC_STOP);
 		if (p->lastinput != AST_FORMAT_G723_1) {
 			p->lastinput = AST_FORMAT_G723_1;
-			if (ioctl(p->fd, IXJCTL_REC_CODEC, G723_63)) {
+			if (ioctl(p->fd, PHONE_REC_CODEC, G723_63)) {
 				ast_log(LOG_WARNING, "Failed to set codec to g723.1\n");
 				return -1;
 			}
 		}
 	} else if (ast->format & AST_FORMAT_SLINEAR) {
-		ioctl(p->fd, IXJCTL_REC_STOP);
+		ioctl(p->fd, PHONE_REC_STOP);
 		if (p->lastinput != AST_FORMAT_SLINEAR) {
 			p->lastinput = AST_FORMAT_SLINEAR;
-			if (ioctl(p->fd, IXJCTL_REC_CODEC, LINEAR16)) {
+			if (ioctl(p->fd, PHONE_REC_CODEC, LINEAR16)) {
 				ast_log(LOG_WARNING, "Failed to set codec to signed linear 16\n");
 				return -1;
 			}
@@ -207,23 +229,33 @@ static int ixj_setup(struct ast_channel *ast)
 		ast_log(LOG_WARNING, "Can't do format %d\n", ast->format);
 		return -1;
 	}
-	if (ioctl(p->fd, IXJCTL_REC_START)) {
+	if (ioctl(p->fd, PHONE_REC_START)) {
 		ast_log(LOG_WARNING, "Failed to start recording\n");
 		return -1;
 	}
 	return 0;
 }
 
-static int ixj_answer(struct ast_channel *ast)
+static int phone_answer(struct ast_channel *ast)
 {
-	ixj_setup(ast);
-	ast_log(LOG_DEBUG, "ixj_answer(%s)\n", ast->name);
+	struct phone_pvt *p;
+	p = ast->pvt->pvt;
+	/* In case it's a LineJack, take it off hook */
+	if (p->mode == MODE_FXO) {
+		if (ioctl(p->fd, PHONE_PSTN_SET_STATE, PSTN_OFF_HOOK)) 
+			ast_log(LOG_DEBUG, "ioctl(PHONE_PSTN_SET_STATE) failed on %s (%s)\n", ast->name, strerror(errno));
+		else
+			ast_log(LOG_DEBUG, "Took linejack off hook\n");
+	}
+	phone_setup(ast);
+	if (option_debug)
+		ast_log(LOG_DEBUG, "phone_answer(%s)\n", ast->name);
 	ast->rings = 0;
 	ast->state = AST_STATE_UP;
 	return 0;
 }
 
-static char ixj_2digit(char c)
+static char phone_2digit(char c)
 {
 	if (c == 12)
 		return '#';
@@ -235,11 +267,11 @@ static char ixj_2digit(char c)
 		return '?';
 }
 
-static struct ast_frame  *ixj_read(struct ast_channel *ast)
+static struct ast_frame  *phone_read(struct ast_channel *ast)
 {
 	int res;
-	IXJ_EXCEPTION ixje;
-	struct ixj_pvt *p = ast->pvt->pvt;
+	union telephony_exception phonee;
+	struct phone_pvt *p = ast->pvt->pvt;
 	char digit;
 
 	/* Some nice norms */
@@ -250,47 +282,75 @@ static struct ast_frame  *ixj_read(struct ast_channel *ast)
 	p->fr.offset = 0;
 	p->fr.mallocd=0;
 
-	ixje.bytes = ioctl(p->fd, IXJCTL_EXCEPTION);
-	if (ixje.bits.dtmf_ready)  {
+	phonee.bytes = ioctl(p->fd, PHONE_EXCEPTION);
+	if (phonee.bits.dtmf_ready)  {
+		if (option_debug)
+			ast_log(LOG_DEBUG, "phone_read(): DTMF\n");
+	
 		/* We've got a digit -- Just handle this nicely and easily */
-		digit =  ioctl(p->fd, IXJCTL_GET_DTMF_ASCII);
+		digit =  ioctl(p->fd, PHONE_GET_DTMF_ASCII);
 		p->fr.subclass = digit;
 		p->fr.frametype = AST_FRAME_DTMF;
 		return &p->fr;
 	}
-	if (ixje.bits.hookstate) {
-		res = ioctl(p->fd, IXJCTL_HOOKSTATE);
+	if (phonee.bits.hookstate) {
+		if (option_debug)
+			ast_log(LOG_DEBUG, "Hookstate changed\n");
+		res = ioctl(p->fd, PHONE_HOOKSTATE);
 		/* See if we've gone on hook, if so, notify by returning NULL */
-		if (!res)
+		if (option_debug)
+			ast_log(LOG_DEBUG, "New hookstate: %d\n", res);
+		if (!res && (p->mode != MODE_FXO))
 			return NULL;
 		else {
 			if (ast->state == AST_STATE_RINGING) {
 				/* They've picked up the phone */
 				p->fr.frametype = AST_FRAME_CONTROL;
 				p->fr.subclass = AST_CONTROL_ANSWER;
-				ixj_setup(ast);
+				phone_setup(ast);
 				ast->state = AST_STATE_UP;
 				return &p->fr;
 			}  else 
-				ast_log(LOG_WARNING, "Got off hook in weird state\n");
+				ast_log(LOG_WARNING, "Got off hook in weird state %d\n", ast->state);
 		}
 	}
-#if 0
-	if (ixje.bits.pstn_ring)
+#if 1
+	if (phonee.bits.pstn_ring)
 		ast_verbose("Unit is ringing\n");
-	if (ixje.bits.caller_id) {
+	if (phonee.bits.caller_id) {
 		ast_verbose("We have caller ID: %s\n");
 	}
+	if (phonee.bits.pstn_wink)
+		ast_verbose("Detected Wink\n");
 #endif
 	/* Try to read some data... */
 	CHECK_BLOCKING(ast);
-	res = read(p->fd, p->buf, IXJ_MAX_BUF);
+	res = read(p->fd, p->buf, PHONE_MAX_BUF);
 	ast->blocking = 0;
 	if (res < 0) {
+#if 0
+		if (errno == EAGAIN) {
+			ast_log(LOG_WARNING, "Null frame received\n");
+			p->fr.frametype = AST_FRAME_NULL;
+			p->fr.subclass = 0;
+			return &p->fr;
+		}
+#endif
 		ast_log(LOG_WARNING, "Error reading: %s\n", strerror(errno));
 		return NULL;
 	}
 	p->fr.data = p->buf;
+	switch(p->buf[0] & 0x3) {
+	case '0':
+	case '1':
+		/* Normal */
+		break;
+	case '2':
+	case '3':
+		/* VAD/CNG, only send two words */
+		res = 4;
+		break;
+	}
 	p->fr.datalen = res;
 	p->fr.frametype = AST_FRAME_VOICE;
 	p->fr.subclass = p->lastinput;
@@ -298,7 +358,7 @@ static struct ast_frame  *ixj_read(struct ast_channel *ast)
 	return &p->fr;
 }
 
-static int ixj_write_buf(struct ixj_pvt *p, char *buf, int len, int frlen)
+static int phone_write_buf(struct phone_pvt *p, char *buf, int len, int frlen)
 {
 	int res;
 	/* Store as much of the buffer as we can, then write fixed frames */
@@ -331,14 +391,16 @@ static int ixj_write_buf(struct ixj_pvt *p, char *buf, int len, int frlen)
 	return len;
 }
 
-static int ixj_write(struct ast_channel *ast, struct ast_frame *frame)
+static int phone_write(struct ast_channel *ast, struct ast_frame *frame)
 {
-	struct ixj_pvt *p = ast->pvt->pvt;
+	struct phone_pvt *p = ast->pvt->pvt;
 	int res;
 	int maxfr=0;
 	char *pos;
 	int sofar;
 	int expected;
+	int codecset = 0;
+	char tmpbuf[4];
 	/* Write a frame of (presumably voice) data */
 	if (frame->frametype != AST_FRAME_VOICE) {
 		ast_log(LOG_WARNING, "Don't know what to do with  frame type '%d'\n", frame->frametype);
@@ -350,16 +412,28 @@ static int ixj_write(struct ast_channel *ast, struct ast_frame *frame)
 		ast_frfree(frame);
 		return -1;
 	}
+	/* If we're not in up mode, go into up mode now */
+	if (ast->state != AST_STATE_UP) {
+		ast->state = AST_STATE_UP;
+		phone_setup(ast);
+	}
 	if (frame->subclass == AST_FORMAT_G723_1) {
 		if (p->lastformat != AST_FORMAT_G723_1) {
-			ioctl(p->fd, IXJCTL_PLAY_STOP);
-			if (ioctl(p->fd, IXJCTL_PLAY_CODEC, G723_63)) {
+			ioctl(p->fd, PHONE_PLAY_STOP);
+			ioctl(p->fd, PHONE_REC_STOP);
+			if (ioctl(p->fd, PHONE_PLAY_CODEC, G723_63)) {
+				ast_log(LOG_WARNING, "Unable to set G723.1 mode\n");
+				return -1;
+			}
+			if (ioctl(p->fd, PHONE_REC_CODEC, G723_63)) {
 				ast_log(LOG_WARNING, "Unable to set G723.1 mode\n");
 				return -1;
 			}
 			p->lastformat = AST_FORMAT_G723_1;
+			p->lastinput = AST_FORMAT_G723_1;
 			/* Reset output buffer */
 			p->obuflen = 0;
+			codecset = 1;
 		}
 		if (frame->datalen > 24) {
 			ast_log(LOG_WARNING, "Frame size too large for G.723.1 (%d bytes)\n", frame->datalen);
@@ -368,20 +442,35 @@ static int ixj_write(struct ast_channel *ast, struct ast_frame *frame)
 		maxfr = 24;
 	} else if (frame->subclass == AST_FORMAT_SLINEAR) {
 		if (p->lastformat != AST_FORMAT_SLINEAR) {
-			ioctl(p->fd, IXJCTL_PLAY_STOP);
-			if (ioctl(p->fd, IXJCTL_PLAY_CODEC, LINEAR16)) {
+			ioctl(p->fd, PHONE_PLAY_STOP);
+			ioctl(p->fd, PHONE_REC_STOP);
+			if (ioctl(p->fd, PHONE_PLAY_CODEC, LINEAR16)) {
+				ast_log(LOG_WARNING, "Unable to set 16-bit linear mode\n");
+				return -1;
+			}
+			if (ioctl(p->fd, PHONE_REC_CODEC, LINEAR16)) {
 				ast_log(LOG_WARNING, "Unable to set 16-bit linear mode\n");
 				return -1;
 			}
 			p->lastformat = AST_FORMAT_SLINEAR;
+			p->lastinput = AST_FORMAT_SLINEAR;
+			codecset = 1;
 			/* Reset output buffer */
 			p->obuflen = 0;
 		}
 		maxfr = 480;
 	}
-	if (ioctl(p->fd, IXJCTL_PLAY_START)) {
-		ast_log(LOG_WARNING, "Failed to start recording\n");
-		return -1;
+	if (codecset) {
+		ioctl(p->fd, PHONE_REC_DEPTH, 3);
+		ioctl(p->fd, PHONE_PLAY_DEPTH, 3);
+		if (ioctl(p->fd, PHONE_PLAY_START)) {
+			ast_log(LOG_WARNING, "Failed to start playback\n");
+			return -1;
+		}
+		if (ioctl(p->fd, PHONE_REC_START)) {
+			ast_log(LOG_WARNING, "Failed to start recording\n");
+			return -1;
+		}
 	}
 	/* If we get here, we have a voice frame of Appropriate data */
 	sofar = 0;
@@ -391,55 +480,67 @@ static int ixj_write(struct ast_channel *ast, struct ast_frame *frame)
 		expected = frame->datalen - sofar;
 		if (maxfr < expected)
 			expected = maxfr;
-		/* XXX Internet Phone Jack does not handle the 4-byte VAD frame properly! XXX */
-		if (frame->datalen != 4) {
-			res = ixj_write_buf(p, pos, expected, maxfr);
-			if (res != expected) {
-				if (res < 0) 
-					ast_log(LOG_WARNING, "Write returned error (%s)\n", strerror(errno));
-				else
-					ast_log(LOG_WARNING, "Only wrote %d of %d bytes\n", res, frame->datalen);
-				return -1;
+		/* XXX Internet Phone Jack does not handle the 4-byte VAD frame properly! XXX 
+		   we have to pad it to 24 bytes still.  */
+		if (frame->datalen == 4) {
+			if (p->silencesupression) {
+				memset(tmpbuf + 4, 0, sizeof(tmpbuf) - 4);
+				memcpy(tmpbuf, frame->data, 4);
+				expected = 24;
+				res = phone_write_buf(p, tmpbuf, expected, maxfr);
 			}
-			sofar += res;
-			pos += res;
-		} else
-			sofar += 4;
+			res = 4;
+			expected=4;
+		} else {
+			res = phone_write_buf(p, pos, expected, maxfr);
+		}
+		if (res != expected) {
+			if (res < 0) 
+				ast_log(LOG_WARNING, "Write returned error (%s)\n", strerror(errno));
+			else
+				ast_log(LOG_WARNING, "Only wrote %d of %d bytes\n", res, frame->datalen);
+			return -1;
+		}
+		sofar += res;
+		pos += res;
 	}
 	return 0;
 }
 
-static struct ast_channel *ixj_new(struct ixj_pvt *i, int state)
+static struct ast_channel *phone_new(struct phone_pvt *i, int state, char *context)
 {
 	struct ast_channel *tmp;
 	tmp = ast_channel_alloc();
 	if (tmp) {
-		snprintf(tmp->name, sizeof(tmp->name), "PhoneJack/%s", i->dev + 5);
+		snprintf(tmp->name, sizeof(tmp->name), "Phone/%s", i->dev + 5);
 		tmp->type = type;
 		tmp->fd = i->fd;
 		/* XXX Switching formats silently causes kernel panics XXX */
-		tmp->format = AST_FORMAT_G723_1 /* | AST_FORMAT_SLINEAR */;
+		tmp->format = prefformat;
 		tmp->state = state;
 		if (state == AST_STATE_RING)
 			tmp->rings = 1;
 		tmp->pvt->pvt = i;
-		tmp->pvt->send_digit = ixj_digit;
-		tmp->pvt->call = ixj_call;
-		tmp->pvt->hangup = ixj_hangup;
-		tmp->pvt->answer = ixj_answer;
-		tmp->pvt->read = ixj_read;
-		tmp->pvt->write = ixj_write;
-		strncpy(tmp->context, i->context, sizeof(tmp->context));
+		tmp->pvt->send_digit = phone_digit;
+		tmp->pvt->call = phone_call;
+		tmp->pvt->hangup = phone_hangup;
+		tmp->pvt->answer = phone_answer;
+		tmp->pvt->read = phone_read;
+		tmp->pvt->write = phone_write;
+		strncpy(tmp->context, context, sizeof(tmp->context));
 		if (strlen(i->ext))
 			strncpy(tmp->exten, i->ext, sizeof(tmp->exten));
+		if (strlen(i->language))
+			strncpy(tmp->language, i->language, sizeof(tmp->language));
 		i->owner = tmp;
 		pthread_mutex_lock(&usecnt_lock);
 		usecnt++;
 		pthread_mutex_unlock(&usecnt_lock);
 		ast_update_use_count();
 		if (state != AST_STATE_DOWN) {
-			if (state == AST_STATE_RING)
-				ioctl(tmp->fd, IXJCTL_RINGBACK);
+			if (state == AST_STATE_RING) {
+				ioctl(tmp->fd, PHONE_RINGBACK);
+			}
 			if (ast_pbx_start(tmp)) {
 				ast_log(LOG_WARNING, "Unable to start PBX on %s\n", tmp->name);
 				ast_hangup(tmp);
@@ -450,7 +551,7 @@ static struct ast_channel *ixj_new(struct ixj_pvt *i, int state)
 	return tmp;
 }
 
-static void ixj_mini_packet(struct ixj_pvt *i)
+static void phone_mini_packet(struct phone_pvt *i)
 {
 	int res;
 	char buf[1024];
@@ -462,78 +563,97 @@ static void ixj_mini_packet(struct ixj_pvt *i)
 	}
 }
 
-static void ixj_check_exception(struct ixj_pvt *i)
+static void phone_check_exception(struct phone_pvt *i)
 {
 	int offhook=0;
 	char digit[2] = {0 , 0};
-	IXJ_EXCEPTION ixje;
+	union telephony_exception phonee;
 	/* XXX Do something XXX */
 #if 0
 	ast_log(LOG_DEBUG, "Exception!\n");
 #endif
-	ixje.bytes = ioctl(i->fd, IXJCTL_EXCEPTION);
-	if (ixje.bits.dtmf_ready)  {
-		digit[0] = ioctl(i->fd, IXJCTL_GET_DTMF_ASCII);
+	phonee.bytes = ioctl(i->fd, PHONE_EXCEPTION);
+	if (phonee.bits.dtmf_ready)  {
+		digit[0] = ioctl(i->fd, PHONE_GET_DTMF_ASCII);
 		if (i->mode == MODE_DIALTONE) {
-			ioctl(i->fd, IXJCTL_PLAY_STOP);
-			ioctl(i->fd, IXJCTL_REC_STOP);
-			ioctl(i->fd, IXJCTL_CPT_STOP);
+			ioctl(i->fd, PHONE_PLAY_STOP);
+			ioctl(i->fd, PHONE_REC_STOP);
+			ioctl(i->fd, PHONE_CPT_STOP);
 			i->dialtone = 0;
 			if (strlen(i->ext) < AST_MAX_EXTENSION - 1)
 				strcat(i->ext, digit);
 			if (ast_exists_extension(NULL, i->context, i->ext, 1)) {
 				/* It's a valid extension in its context, get moving! */
-				ixj_new(i, AST_STATE_UP);
+				phone_new(i, AST_STATE_RING, i->context);
 				/* No need to restart monitor, we are the monitor */
 				if (i->owner) {
-					ixj_setup(i->owner);
+					pthread_mutex_lock(&usecnt_lock);
+					usecnt--;
+					pthread_mutex_unlock(&usecnt_lock);
+					ast_update_use_count();
 				}
-			} else if (ast_exists_extension(NULL, "default", i->ext, 1)) {
-				/* Check the default, too... */
-				/* XXX This should probably be justified better XXX */
-				strncpy(i->context, "default", sizeof(i->context));
-				ixj_new(i, AST_STATE_UP);
-				if (i->owner) {
-					ixj_setup(i->owner);
+			} else if (!ast_canmatch_extension(NULL, i->context, i->ext, 1)) {
+				/* There is nothing in the specified extension that can match anymore.
+				   Try the default */
+				if (ast_exists_extension(NULL, "default", i->ext, 1)) {
+					/* Check the default, too... */
+					phone_new(i, AST_STATE_RING, "default");
+					if (i->owner) {
+						pthread_mutex_lock(&usecnt_lock);
+						usecnt--;
+						pthread_mutex_unlock(&usecnt_lock);
+						ast_update_use_count();
+					}
+					/* XXX This should probably be justified better XXX */
+				}  else if (!ast_canmatch_extension(NULL, "default", i->ext, 1)) {
+					/* It's not a valid extension, give a busy signal */
+					if (option_debug)
+						ast_log(LOG_DEBUG, "%s can't match anything in %s or default\n", i->ext, i->context);
+					ioctl(i->fd, PHONE_BUSY);
 				}
-			} else if ((strlen(i->ext) >= ast_pbx_longest_extension(i->context)) &&
-					   (strlen(i->ext) >= ast_pbx_longest_extension("default"))) {
-				/* It's not a valid extension, give a busy signal */
-				ioctl(i->fd, IXJCTL_BUSY);
 			}
 #if 0
 			ast_verbose("Extension is %s\n", i->ext);
 #endif
 		}
 	}
-	if (ixje.bits.hookstate) {
-		offhook = ioctl(i->fd, IXJCTL_HOOKSTATE);
+	if (phonee.bits.hookstate) {
+		offhook = ioctl(i->fd, PHONE_HOOKSTATE);
 		if (offhook) {
 			if (i->mode == MODE_IMMEDIATE) {
-				ixj_new(i, AST_STATE_RING);
+				phone_new(i, AST_STATE_RING, i->context);
 			} else if (i->mode == MODE_DIALTONE) {
-#if 0
-				/* XXX Bug in the Phone jack, you can't detect DTMF when playing a tone XXX */
-				ioctl(i->fd, IXJCTL_DIALTONE);
-#else
+				pthread_mutex_lock(&usecnt_lock);
+				usecnt++;
+				pthread_mutex_unlock(&usecnt_lock);
+				ast_update_use_count();
+				/* Reset the extension */
+				i->ext[0] = '\0';
 				/* Play the dialtone */
 				i->dialtone++;
-				ioctl(i->fd, IXJCTL_PLAY_STOP);
-				ioctl(i->fd, IXJCTL_PLAY_CODEC, ULAW);
-				ioctl(i->fd, IXJCTL_PLAY_START);
-#endif
+				ioctl(i->fd, PHONE_PLAY_STOP);
+				ioctl(i->fd, PHONE_PLAY_CODEC, ULAW);
+				ioctl(i->fd, PHONE_PLAY_START);
 			}
 		} else {
+			if (i->dialtone) {
+				pthread_mutex_lock(&usecnt_lock);
+				usecnt--;
+				pthread_mutex_unlock(&usecnt_lock);
+				ast_update_use_count();
+			}
 			memset(i->ext, 0, sizeof(i->ext));
-			ioctl(i->fd, IXJCTL_CPT_STOP);
-			ioctl(i->fd, IXJCTL_PLAY_STOP);
-			ioctl(i->fd, IXJCTL_REC_STOP);
+			ioctl(i->fd, PHONE_CPT_STOP);
+			ioctl(i->fd, PHONE_PLAY_STOP);
+			ioctl(i->fd, PHONE_REC_STOP);
 			i->dialtone = 0;
 		}
 	}
-	if (ixje.bits.pstn_ring)
+	if (phonee.bits.pstn_ring) {
 		ast_verbose("Unit is ringing\n");
-	if (ixje.bits.caller_id)
+		phone_new(i, AST_STATE_RING, i->context);
+	}
+	if (phonee.bits.caller_id)
 		ast_verbose("We have caller ID\n");
 	
 	
@@ -543,7 +663,7 @@ static void *do_monitor(void *data)
 {
 	fd_set rfds, efds;
 	int n, res;
-	struct ixj_pvt *i;
+	struct phone_pvt *i;
 	int tonepos = 0;
 	/* The tone we're playing this round */
 	struct timeval tv = {0,0};
@@ -569,7 +689,7 @@ static void *do_monitor(void *data)
 			return NULL;
 		}
 		/* Build the stuff we're going to select on, that is the socket of every
-		   ixj_pvt that does not have an associated owner channel */
+		   phone_pvt that does not have an associated owner channel */
 		n = -1;
 		FD_ZERO(&rfds);
 		FD_ZERO(&efds);
@@ -642,14 +762,14 @@ static void *do_monitor(void *data)
 					ast_log(LOG_WARNING, "Whoa....  I'm owned but found (%d, %s)...\n", i->fd, i->dev);
 					continue;
 				}
-				ixj_mini_packet(i);
+				phone_mini_packet(i);
 			}
 			if (FD_ISSET(i->fd, &efds)) {
 				if (i->owner) {
 					ast_log(LOG_WARNING, "Whoa....  I'm owned but found (%d, %s)...\n", i->fd, i->dev);
 					continue;
 				}
-				ixj_check_exception(i);
+				phone_check_exception(i);
 			}
 			i=i->next;
 		}
@@ -690,13 +810,15 @@ static int restart_monitor()
 	return 0;
 }
 
-struct ixj_pvt *mkif(char *iface, int mode)
+static struct phone_pvt *mkif(char *iface, int mode)
 {
-	/* Make a ixj_pvt structure for this interface */
-	struct ixj_pvt *tmp;
+	/* Make a phone_pvt structure for this interface */
+	struct phone_pvt *tmp;
+#if 0
 	int flags;	
+#endif
 	
-	tmp = malloc(sizeof(struct ixj_pvt));
+	tmp = malloc(sizeof(struct phone_pvt));
 	if (tmp) {
 		tmp->fd = open(iface, O_RDWR);
 		if (tmp->fd < 0) {
@@ -704,18 +826,34 @@ struct ixj_pvt *mkif(char *iface, int mode)
 			free(tmp);
 			return NULL;
 		}
-		ioctl(tmp->fd, IXJCTL_PLAY_STOP);
-		ioctl(tmp->fd, IXJCTL_REC_STOP);
-		ioctl(tmp->fd, IXJCTL_RING_STOP);
-		ioctl(tmp->fd, IXJCTL_CPT_STOP);
+		if (mode == MODE_FXO) {
+			if (ioctl(tmp->fd, IXJCTL_PORT, PORT_PSTN)) 
+				ast_log(LOG_DEBUG, "Unable to set port to PSTN\n");
+		}
+		ioctl(tmp->fd, PHONE_PLAY_STOP);
+		ioctl(tmp->fd, PHONE_REC_STOP);
+		ioctl(tmp->fd, PHONE_RING_STOP);
+		ioctl(tmp->fd, PHONE_CPT_STOP);
+		if (ioctl(tmp->fd, PHONE_PSTN_SET_STATE, PSTN_ON_HOOK)) 
+			ast_log(LOG_DEBUG, "ioctl(PHONE_PSTN_SET_STATE) failed on %s (%s)\n",iface, strerror(errno));
+		if (echocancel != AEC_OFF)
+			ioctl(tmp->fd, IXJCTL_AEC_START, echocancel);
+		if (silencesupression) 
+			tmp->silencesupression = 1;
+#ifdef PHONE_VAD
+		ioctl(tmp->fd, PHONE_VAD, tmp->silencesupression);
+#endif
 		tmp->mode = mode;
+#if 0
 		flags = fcntl(tmp->fd, F_GETFL);
 		fcntl(tmp->fd, F_SETFL, flags | O_NONBLOCK);
+#endif
 		tmp->owner = NULL;
 		tmp->lastformat = -1;
 		tmp->lastinput = -1;
 		tmp->ministate = 0;
 		memset(tmp->ext, 0, sizeof(tmp->ext));
+		strncpy(tmp->language, language, sizeof(tmp->language));
 		strncpy(tmp->dev, iface, sizeof(tmp->dev));
 		strncpy(tmp->context, context, sizeof(tmp->context));
 		tmp->next = NULL;
@@ -725,10 +863,10 @@ struct ixj_pvt *mkif(char *iface, int mode)
 	return tmp;
 }
 
-static struct ast_channel *ixj_request(char *type, int format, void *data)
+static struct ast_channel *phone_request(char *type, int format, void *data)
 {
 	int oldformat;
-	struct ixj_pvt *p;
+	struct phone_pvt *p;
 	struct ast_channel *tmp = NULL;
 	char *name = data;
 	
@@ -747,7 +885,7 @@ static struct ast_channel *ixj_request(char *type, int format, void *data)
 	while(p) {
 		if (!strcmp(name, p->dev + 5)) {
 			if (!p->owner) {
-				tmp = ixj_new(p, AST_STATE_DOWN);
+				tmp = phone_new(p, AST_STATE_DOWN, p->context);
 				break;
 			}
 		}
@@ -762,7 +900,7 @@ int load_module()
 {
 	struct ast_config *cfg;
 	struct ast_variable *v;
-	struct ixj_pvt *tmp;
+	struct phone_pvt *tmp;
 	int mode = MODE_IMMEDIATE;
 	cfg = ast_load(config);
 
@@ -792,21 +930,45 @@ int load_module()
 					unload_module();
 					return -1;
 				}
+		} else if (!strcasecmp(v->name, "silencesupression")) {
+			silencesupression = ast_true(v->value);
+		} else if (!strcasecmp(v->name, "language")) {
+			strncpy(language, v->value, sizeof(language));
 		} else if (!strcasecmp(v->name, "mode")) {
 			if (!strncasecmp(v->value, "di", 2)) 
 				mode = MODE_DIALTONE;
 			else if (!strncasecmp(v->value, "im", 2))
 				mode = MODE_IMMEDIATE;
+			else if (!strncasecmp(v->value, "fx", 2))
+				mode = MODE_FXO;
 			else
 				ast_log(LOG_WARNING, "Unknown mode: %s\n", v->value);
 		} else if (!strcasecmp(v->name, "context")) {
 			strncpy(context, v->value, sizeof(context));
+		} else if (!strcasecmp(v->name, "format")) {
+			if (!strcasecmp(v->value, "g723.1")) {
+				prefformat = AST_FORMAT_G723_1;
+			} else if (!strcasecmp(v->value, "slinear")) {
+				prefformat = AST_FORMAT_SLINEAR;
+			} else 
+				ast_log(LOG_WARNING, "Unknown format '%s'\n", v->value);
+		} else if (!strcasecmp(v->name, "echocancel")) {
+			if (!strcasecmp(v->value, "off")) {
+				echocancel = AEC_OFF;
+			} else if (!strcasecmp(v->value, "low")) {
+				echocancel = AEC_LOW;
+			} else if (!strcasecmp(v->value, "medium")) {
+				echocancel = AEC_MED;
+			} else if (!strcasecmp(v->value, "high")) {
+				echocancel = AEC_HIGH;
+			} else 
+				ast_log(LOG_WARNING, "Unknown echo cancellation '%s'\n", v->value);
 		}
 		v = v->next;
 	}
 	pthread_mutex_unlock(&iflock);
-	/* Make sure we can register our Adtranixj channel type */
-	if (ast_channel_register(type, tdesc, AST_FORMAT_G723_1, ixj_request)) {
+	/* Make sure we can register our Adtranphone channel type */
+	if (ast_channel_register(type, tdesc, AST_FORMAT_G723_1, phone_request)) {
 		ast_log(LOG_ERROR, "Unable to register channel class %s\n", type);
 		ast_destroy(cfg);
 		unload_module();
@@ -822,7 +984,7 @@ int load_module()
 
 int unload_module()
 {
-	struct ixj_pvt *p, *pl;
+	struct phone_pvt *p, *pl;
 	/* First, take us out of the channel loop */
 	ast_channel_unregister(type);
 	if (!pthread_mutex_lock(&iflock)) {

channels/chan_vofr.c

@@ -3,7 +3,7 @@
  *
  * Implementation of Voice over Frame Relay, Adtran Style
  * 
- * Copyright (C) 1999, Adtran Inc. and Linux Support Services, LLC
+ * Copyright (C) 1999, Mark Spencer
  *
  * Mark Spencer <markster@linux-support.net>
  *
@@ -25,11 +25,19 @@
 #include <errno.h>
 #include <unistd.h>
 #include <stdlib.h>
+#include <fcntl.h>
 #include <arpa/inet.h>
 #include <linux/if_packet.h>
 #include <linux/if_ether.h>
+#ifndef OLD_SANGOMA_API
+#include <linux/if_wanpipe.h>
+#include <linux/wanpipe.h>
+#endif
+#include <sys/signal.h>
 #include "adtranvofr.h"
 
+/* #define VOFRDUMPER */
+
 #define G723_MAX_BUF 2048
 
 #define FR_API_MESS 16
@@ -41,6 +49,8 @@ static char *config = "adtranvofr.conf";
 
 static char context[AST_MAX_EXTENSION] = "default";
 
+static char language[MAX_LANGUAGE] = "";
+
 static int usecnt =0;
 static pthread_mutex_t usecnt_lock = PTHREAD_MUTEX_INITIALIZER;
 
@@ -53,16 +63,20 @@ static pthread_mutex_t monlock = PTHREAD_MUTEX_INITIALIZER;
 
 /* This is the thread for the monitor which checks for input on the channels
    which are not currently in use.  */
-static pthread_t monitor_thread = -1;
+static pthread_t monitor_thread = 0;
 
-static int restart_monitor();
+static int restart_monitor(void);
 
 /* The private structures of the Adtran VoFR channels are linked for
    selecting outgoing channels */
    
 static struct vofr_pvt {
 	int s;							/* Raw socket for this DLCI */
+#ifdef OLD_SANGOMA_API
 	struct sockaddr_pkt sa;			/* Sockaddr needed for sending, also has iface name */
+#else
+	struct wan_sockaddr_ll sa;		/* Wanpipe sockaddr */
+#endif
 	struct ast_channel *owner;		/* Channel we belong to, possibly NULL */
 	int outgoing;					/* Does this channel support outgoing calls? */
 	struct vofr_pvt *next;			/* Next channel in list */
@@ -74,6 +88,8 @@ static struct vofr_pvt {
 	char buf[G723_MAX_BUF];					/* Static buffer for reading frames */
 	char obuf[G723_MAX_BUF];				/* Output buffer */
 	char context[AST_MAX_EXTENSION];
+	char language[MAX_LANGUAGE];
+	int ringgothangup;				/* Have we received exactly one hangup after a ring */
 } *iflist = NULL;
 
 #ifdef VOFRDUMPER
@@ -240,7 +256,11 @@ static void vofr_dump_packet(struct vofr_hdr *vh, int len)
 static int vofr_xmit(struct vofr_pvt *p, char *data, int len)
 {
 	int res;
+#ifdef OLD_SANGOMA_API
     res=sendto(p->s, data, len, 0, (struct sockaddr *)&p->sa, sizeof(struct sockaddr_pkt));
+#else
+    res=sendto(p->s, data, len, 0, (struct sockaddr *)&p->sa, sizeof(struct wan_sockaddr_ll));
+#endif
 	if (res != len) {
 		ast_log(LOG_WARNING, "vofr_xmit returned %d\n", res);
 	}
@@ -336,7 +356,7 @@ static int vofr_call(struct ast_channel *ast, char *dest, int timeout)
 	while(otimeout) {
 		otimeout = ast_waitfor(ast, 1000);
 		if (otimeout < 1) {
-			ast_log(LOG_WARNING, "Unable to take line off hook\n");
+			ast_log(LOG_WARNING, "Unable to take line '%s' off hook\n", ast->name);
 			/* Musta gotten hung up, or no ack on off hook */
 			return -1;	
 		}
@@ -433,6 +453,7 @@ static int vofr_hangup(struct ast_channel *ast)
 	}
 	ast->state = AST_STATE_DOWN;
 	((struct vofr_pvt *)(ast->pvt->pvt))->owner = NULL;
+	((struct vofr_pvt *)(ast->pvt->pvt))->ringgothangup = 0;
 	pthread_mutex_lock(&usecnt_lock);
 	usecnt--;
 	if (usecnt < 0) 
@@ -464,6 +485,9 @@ static int vofr_answer(struct ast_channel *ast)
 		cnt = ast_waitfor(ast, cnt);
 		if (cnt > 0) {
 			res = read(ast->fd, buf, sizeof(buf));
+#ifdef VOFRDUMPER
+				vofr_dump_packet((void *)(buf +FR_API_MESS), res - FR_API_MESS);
+#endif
 			res -= FR_API_MESS;
 			if (res < 0)
 				ast_log(LOG_WARNING, "Warning:  read failed (%s) on %s\n", strerror(errno), ast->name);
@@ -528,15 +552,32 @@ static struct ast_frame  *vofr_read(struct ast_channel *ast)
 	/* Read into the right place in the buffer, in case we send this
 	   as a voice frame. */
 	CHECK_BLOCKING(ast);
+retry:
 	res = read(p->s, ((char *)vh)  - FR_API_MESS, 
 				G723_MAX_BUF - AST_FRIENDLY_OFFSET - sizeof(struct ast_frame) + sizeof(struct vofr_hdr) + FR_API_MESS);
+	if (res < 0) {
+		/*  XXX HUGE BUG IN SANGOMA'S STACK: IT IGNORES O_NONBLOCK XXX */
+		if (errno == EAGAIN) {
+			fd_set fds;
+			FD_ZERO(&fds);
+			FD_SET(p->s, &fds);
+			select(p->s + 1, &fds, NULL, NULL, NULL);
+			goto retry;
+		}
+		ast->blocking = 0;
+		ast_log(LOG_WARNING, "Read error on %s: %s (%d)\n", ast->name, strerror(errno));
+		return NULL;
+	}
 	ast->blocking = 0;
+		
+#ifdef VOFRDUMPER
+	vofr_dump_packet((void *)(vh), res);
+#endif
 	res -= FR_API_MESS;		
-	if (res < sizeof(struct vofr_hdr *)) {
+	if (res < sizeof(struct vofr_hdr)) {
 		ast_log(LOG_WARNING, "Nonsense frame on %s\n", ast->name);
 		return NULL;
 	}
-
 	/* Some nice norms */
 	fr->datalen = 0;
 	fr->timelen = 0;
@@ -551,13 +592,18 @@ static struct ast_frame  *vofr_read(struct ast_channel *ast)
 		switch(vh->data[0]) {
 		case VOFR_SIGNAL_ON_HOOK:
 			/* Hang up this line */
-			if (ast->state == AST_STATE_UP)
+			if ((ast->state == AST_STATE_UP) || (p->ringgothangup)) {
 				return NULL;
-			else {
+			} else {
 				fr->frametype = AST_FRAME_NULL;
 				fr->subclass = 0;
-				break;
+				p->ringgothangup=1;
 			}
+			break;
+		case VOFR_SIGNAL_RING:
+			ast->rings++;
+			p->ringgothangup = 0;
+			break;
 		case VOFR_SIGNAL_UNKNOWN:
 			switch(vh->data[1]) {
 			case 0x1:
@@ -653,7 +699,7 @@ static struct ast_frame  *vofr_read(struct ast_channel *ast)
 			ast->state = AST_STATE_UP;
 			return fr;
 		} else if (ast->state !=  AST_STATE_UP) {
-			ast_log(LOG_WARNING, "Voice in weird state %d\n", ast->state);
+			ast_log(LOG_WARNING, "%s: Voice in weird state %d\n", ast->name, ast->state);
 		}
 		fr->frametype = AST_FRAME_VOICE;
 		fr->subclass = AST_FORMAT_G723_1;
@@ -741,7 +787,11 @@ static struct ast_channel *vofr_new(struct vofr_pvt *i, int state)
 	struct ast_channel *tmp;
 	tmp = ast_channel_alloc();
 	if (tmp) {
+#ifdef OLD_SANGOMA_API
 		snprintf(tmp->name, sizeof(tmp->name), "AdtranVoFR/%s", i->sa.spkt_device);
+#else
+		snprintf(tmp->name, sizeof(tmp->name), "AdtranVoFR/%s", i->sa.sll_device);
+#endif
 		tmp->type = type;
 		tmp->fd = i->s;
 		/* Adtran VoFR supports only G723.1 format data.  G711 (ulaw) would be nice too */
@@ -756,6 +806,8 @@ static struct ast_channel *vofr_new(struct vofr_pvt *i, int state)
 		tmp->pvt->answer = vofr_answer;
 		tmp->pvt->read = vofr_read;
 		tmp->pvt->write = vofr_write;
+		if (strlen(i->language))
+			strncpy(tmp->language, i->language, sizeof(tmp->language));
 		i->owner = tmp;
 		pthread_mutex_lock(&usecnt_lock);
 		usecnt++;
@@ -766,6 +818,7 @@ static struct ast_channel *vofr_new(struct vofr_pvt *i, int state)
 			if (ast_pbx_start(tmp)) {
 				ast_log(LOG_WARNING, "Unable to start PBX on %s\n", tmp->name);
 				ast_hangup(tmp);
+				tmp = NULL;
 			}
 		}
 	} else
@@ -783,7 +836,22 @@ static int vofr_mini_packet(struct vofr_pvt *i, struct vofr_hdr *pkt, int len)
 		switch(pkt->data[0]) {
 		case VOFR_SIGNAL_RING:
 			/* If we get a RING, we definitely want to start a new thread */
-			vofr_new(i, AST_STATE_RING);
+			if (!i->owner) {
+				i->ringgothangup = 0;
+				vofr_new(i, AST_STATE_RING);
+			} else
+				ast_log(LOG_WARNING, "Got a ring, but there's an owner?\n");
+			break;
+		case VOFR_SIGNAL_OFF_HOOK:
+			/* Network termination, go off hook */
+#if 0
+			ast_log(LOG_DEBUG, "Off hook\n");
+#endif
+			vofr_xmit_signal(i, 0x10, 2);
+			if (!i->owner)
+				vofr_new(i, AST_STATE_UP);
+			else
+				ast_log(LOG_WARNING, "Got an offhook, but there's an owner?\n");
 			break;
 		case VOFR_SIGNAL_ON_HOOK:
 			break;
@@ -822,10 +890,12 @@ static void *do_monitor(void *data)
 	/* This thread monitors all the frame relay interfaces which are not yet in use
 	   (and thus do not have a separate thread) indefinitely */
 	/* From here on out, we die whenever asked */
+#if 0
 	if (pthread_setcanceltype(PTHREAD_CANCEL_ASYNCHRONOUS, NULL)) {
 		ast_log(LOG_WARNING, "Unable to set cancel type to asynchronous\n");
 		return NULL;
 	}
+#endif
 	for(;;) {
 		/* Don't let anybody kill us right away.  Nobody should lock the interface list
 		   and wait for the monitor list, but the other way around is okay. */
@@ -846,7 +916,11 @@ static void *do_monitor(void *data)
 		i = iflist;
 		while(i) {
 			if (FD_ISSET(i->s, &rfds)) 
+#ifdef OLD_SANGOMA_API
 				ast_log(LOG_WARNING, "Descriptor %d appears twice (%s)?\n", i->s, i->sa.spkt_device);
+#else
+				ast_log(LOG_WARNING, "Descriptor %d appears twice (%s)?\n", i->s, i->sa.sll_device);
+#endif
 			if (!i->owner) {
 				/* This needs to be watched, as it lacks an owner */
 				FD_SET(i->s, &rfds);
@@ -860,11 +934,14 @@ static void *do_monitor(void *data)
 		
 		/* And from now on, we're okay to be killed, so release the monitor lock as well */
 		pthread_mutex_unlock(&monlock);
+		pthread_testcancel();
 		/* Wait indefinitely for something to happen */
 		res = select(n + 1, &rfds, NULL, NULL, NULL);
+		pthread_testcancel();
 		/* Okay, select has finished.  Let's see what happened.  */
 		if (res < 0) {
-			ast_log(LOG_WARNING, "select return %d: %s\n", res, strerror(errno));
+			if ((errno != EAGAIN) && (errno != EINTR))
+				ast_log(LOG_WARNING, "select return %d: %s\n", res, strerror(errno));
 			continue;
 		}
 		/* Alright, lock the interface list again, and let's look and see what has
@@ -877,7 +954,11 @@ static void *do_monitor(void *data)
 		while(i) {
 			if (FD_ISSET(i->s, &rfds)) {
 				if (i->owner) {
+#ifdef OLD_SANGOMA_API
 					ast_log(LOG_WARNING, "Whoa....  I'm owned but found (%d, %s)...\n", i->s, i->sa.spkt_device);
+#else
+					ast_log(LOG_WARNING, "Whoa....  I'm owned but found (%d, %s)...\n", i->s, i->sa.sll_device);
+#endif
 					continue;
 				}
 				res = read(i->s, i->buf, sizeof(i->buf));
@@ -896,7 +977,7 @@ static void *do_monitor(void *data)
 	
 }
 
-static int restart_monitor()
+static int restart_monitor(void)
 {
 	/* If we're supposed to be stopped -- stay stopped */
 	if (monitor_thread == -2)
@@ -910,12 +991,16 @@ static int restart_monitor()
 		ast_log(LOG_WARNING, "Cannot kill myself\n");
 		return -1;
 	}
-	if (monitor_thread != -1) {
+	if (monitor_thread) {
+#if 0
 		pthread_cancel(monitor_thread);
+#endif
+		pthread_kill(monitor_thread, SIGURG);
 #if 0
 		pthread_join(monitor_thread, NULL);
 #endif
 	}
+	if (!monitor_thread)
 	/* Start a new monitor */
 	if (pthread_create(&monitor_thread, NULL, do_monitor, NULL) < 0) {
 		pthread_mutex_unlock(&monlock);
@@ -926,32 +1011,54 @@ static int restart_monitor()
 	return 0;
 }
 
-struct vofr_pvt *mkif(char *type, char *iface)
+static struct vofr_pvt *mkif(char *type, char *iface)
 {
 	/* Make a vofr_pvt structure for this interface */
 	struct vofr_pvt *tmp;
 	int sndbuf = 4096;
-	
+
 	tmp = malloc(sizeof(struct vofr_pvt));
 	if (tmp) {
 
 		/* Allocate a packet socket */
+#ifdef OLD_SANGOMA_API
 		tmp->s = socket(AF_INET, SOCK_PACKET, htons(ETH_P_ALL));
+#else
+		/* Why the HELL does Sangoma change their API every damn time
+		   they make a new driver release?!?!?!  Leave it the hell
+		   alone this time.  */
+		tmp->s = socket(AF_WANPIPE, SOCK_RAW, 0);
+#endif		
+
 		if (tmp->s < 0) {
 			ast_log(LOG_ERROR, "Unable to create socket: %s\n", strerror(errno));
 			free(tmp);
 			return NULL;
 		}
 
+#ifdef OLD_SANGOMA_API
 		/* Prepare sockaddr for binding */
 		memset(&tmp->sa, 0, sizeof(tmp->sa));
 		strncpy(tmp->sa.spkt_device, iface, sizeof(tmp->sa.spkt_device));
 		tmp->sa.spkt_protocol = htons(0x16);
 		tmp->sa.spkt_family = AF_PACKET;
-		
-		/* Bind socket to specific interface */
 		if (bind(tmp->s, (struct sockaddr *)&tmp->sa, sizeof(struct sockaddr))) {
+#else
+		/* Prepare sockaddr for binding */
+		memset(&tmp->sa, 0, sizeof(tmp->sa));
+		tmp->sa.sll_family = AF_WANPIPE;
+		tmp->sa.sll_protocol = htons(ETH_P_IP);
+		strncpy(tmp->sa.sll_device, iface, sizeof(tmp->sa.sll_device));
+		strncpy(tmp->sa.sll_card, "wanpipe1", sizeof(tmp->sa.sll_card));
+		tmp->sa.sll_ifindex = 0;
+		if (bind(tmp->s, (struct sockaddr *)&tmp->sa, sizeof(struct wan_sockaddr_ll))) {
+#endif		
+		/* Bind socket to specific interface */
+#ifdef OLD_SANGOMA_API
 			ast_log(LOG_ERROR, "Unable to bind to '%s': %s\n", tmp->sa.spkt_device, 
+#else
+			ast_log(LOG_ERROR, "Unable to bind to '%s': %s\n", tmp->sa.sll_device, 
+#endif
 										strerror(errno));
 			free(tmp);
 			return NULL;
@@ -969,6 +1076,8 @@ struct vofr_pvt *mkif(char *type, char *iface)
 		tmp->dlcil = 0;
 		tmp->dlcih = 0;
 		tmp->cid = 1;
+		tmp->ringgothangup = 0;
+		strncpy(tmp->language, language, sizeof(tmp->language));
 		strncpy(tmp->context, context, sizeof(tmp->context));
 		/* User terminations are game for outgoing connections */
 		if (!strcasecmp(type, "user")) 
@@ -1004,7 +1113,7 @@ static struct ast_channel *vofr_request(char *type, int format, void *data)
 	}
 	p = iflist;
 	while(p) {
-		if (!p->owner) {
+		if (!p->owner && p->outgoing) {
 			tmp = vofr_new(p, AST_STATE_DOWN);
 			break;
 		}
@@ -1050,6 +1159,8 @@ int load_module()
 				}
 		} else if (!strcasecmp(v->name, "context")) {
 			strncpy(context, v->value, sizeof(context));
+		} else if (!strcasecmp(v->name, "language")) {
+			strncpy(language, v->value, sizeof(language));
 		}
 		v = v->next;
 	}
@@ -1067,8 +1178,6 @@ int load_module()
 	return 0;
 }
 
-
-
 int unload_module()
 {
 	struct vofr_pvt *p, *pl;
@@ -1089,8 +1198,9 @@ int unload_module()
 		return -1;
 	}
 	if (!pthread_mutex_lock(&monlock)) {
-		if (monitor_thread > -1) {
+		if (monitor_thread) {
 			pthread_cancel(monitor_thread);
+			pthread_kill(monitor_thread, SIGURG);
 			pthread_join(monitor_thread, NULL);
 		}
 		monitor_thread = -2;

channels/iax.h

@@ -0,0 +1,64 @@
+/*
+ * Asterisk -- A telephony toolkit for Linux.
+ *
+ * Implementation of Inter-Asterisk eXchange
+ * 
+ * Copyright (C) 1999, Mark Spencer
+ *
+ * Mark Spencer <markster@linux-support.net>
+ *
+ * This program is free software, distributed under the terms of
+ * the GNU General Public License
+ */
+ 
+#ifndef _ASTERISK_IAX_H
+#define _ASTERISK_IAX_H
+
+/* Max version of IAX protocol we support */
+#define AST_IAX_PROTO_VERSION 1
+
+#define AST_IAX_MAX_CALLS 32768
+
+#define AST_FLAG_FULL		0x8000
+
+#define AST_FLAG_SC_LOG		0x80
+
+#define AST_MAX_SHIFT		0x1F
+
+/* Subclass for AST_FRAME_IAX */
+#define AST_IAX_COMMAND_NEW		1
+#define AST_IAX_COMMAND_PING	2
+#define AST_IAX_COMMAND_PONG	3
+#define AST_IAX_COMMAND_ACK		4
+#define AST_IAX_COMMAND_HANGUP	5
+#define AST_IAX_COMMAND_REJECT	6
+#define AST_IAX_COMMAND_ACCEPT	7
+#define AST_IAX_COMMAND_AUTHREQ	8
+#define AST_IAX_COMMAND_AUTHREP	9
+#define AST_IAX_COMMAND_INVAL	10
+#define AST_IAX_COMMAND_LAGRQ	11
+#define AST_IAX_COMMAND_LAGRP	12
+
+#define AST_DEFAULT_IAX_PORTNO	5036
+
+/* Full frames are always delivered reliably */
+struct ast_iax_full_hdr {
+	short callno;			/* Source call number -- high bit must be 1 */
+	short dcallno;			/* Destination call number */
+	unsigned int ts;		/* 32-bit timestamp in milliseconds */
+	unsigned short seqno;	/* Packet number */
+	char type;				/* Frame type */
+	unsigned char csub;		/* Compressed subclass */
+	char data[0];
+};
+
+/* Mini header is used only for voice frames -- delivered unreliably */
+struct ast_iax_mini_hdr {
+	short callno;			/* Source call number -- high bit must be 0 */
+	unsigned short ts;		/* 16-bit Timestamp (high 16 bits from last ast_iax_full_hdr) */
+							/* Frametype implicitly VOICE_FRAME */
+							/* subclass implicit from last ast_iax_full_hdr */
+	char data[0];
+};
+
+#endif

channels/ixjuser.h

@@ -1,8 +1,8 @@
 /******************************************************************************
 $Id$
 $Log$
-Revision 1.1  1999/12/01 05:25:58  markster
-Version 0.1.0 from FTP
+Revision 1.6  1999/12/01 05:25:58  markster
+Version 0.1.5 from FTP
 
 Revision 1.1  1999/12/01 05:25:58  markster
 Start on the Internet Phone Jack channel

cli.c

@@ -0,0 +1,612 @@
+/*
+ * Asterisk -- A telephony toolkit for Linux.
+ *
+ * Standard Command Line Interface
+ * 
+ * Copyright (C) 1999, Mark Spencer
+ *
+ * Mark Spencer <markster@linux-support.net>
+ *
+ * This program is free software, distributed under the terms of
+ * the GNU General Public License
+ */
+
+#include <unistd.h>
+#include <stdlib.h>
+#include <asterisk/logger.h>
+#include <asterisk/options.h>
+#include <asterisk/cli.h>
+#include <asterisk/module.h>
+#include <asterisk/channel.h>
+#include <sys/signal.h>
+#include <stdio.h>
+#include <signal.h>
+#include <string.h>
+#include <pthread.h>
+/* For rl_filename_completion */
+#include <readline/readline.h>
+/* For module directory */
+#include "asterisk.h"
+
+void ast_cli(int fd, char *fmt, ...)
+{
+	char stuff[4096];
+	va_list ap;
+	va_start(ap, fmt);
+	vsnprintf(stuff, sizeof(stuff), fmt, ap);
+	va_end(ap);
+	write(fd, stuff, strlen(stuff));
+}
+
+pthread_mutex_t clilock = PTHREAD_MUTEX_INITIALIZER;
+
+
+struct ast_cli_entry *helpers = NULL;
+
+static char load_help[] = 
+"Usage: load <module name>\n"
+"       Loads the specified module into Asterisk.\n";
+
+static char unload_help[] = 
+"Usage: unload [-f|-h] <module name>\n"
+"       Unloads the specified module from Asterisk.  The -f\n"
+"       option causes the module to be unloaded even if it is\n"
+"       in use (may cause a crash) and the -h module causes the\n"
+"       module to be unloaded even if the module says it cannot, \n"
+"       which almost always will cause a crash.\n";
+
+static char help_help[] =
+"Usage: help [topic]\n"
+"       When called with a topic as an argument, displays usage\n"
+"       information on the given command.  If called without a\n"
+"       topic, it provides a list of commands.\n";
+
+static char chanlist_help[] = 
+"Usage: show channels\n"
+"       Lists currently defined channels and some information about\n"
+"       them.\n";
+
+static int handle_load(int fd, int argc, char *argv[])
+{
+	if (argc != 2)
+		return RESULT_SHOWUSAGE;
+	if (ast_load_resource(argv[1])) {
+		ast_cli(fd, "Unable to load module %s\n", argv[1]);
+		return RESULT_FAILURE;
+	}
+	return RESULT_SUCCESS;
+}
+
+static int handle_unload(int fd, int argc, char *argv[])
+{
+	int x;
+	int force=AST_FORCE_SOFT;
+	if (argc < 2)
+		return RESULT_SHOWUSAGE;
+	for (x=1;x<argc;x++) {
+		if (argv[x][0] == '-') {
+			switch(argv[x][1]) {
+			case 'f':
+				force = AST_FORCE_FIRM;
+				break;
+			case 'h':
+				force = AST_FORCE_HARD;
+				break;
+			default:
+				return RESULT_SHOWUSAGE;
+			}
+		} else if (x !=  argc - 1) 
+			return RESULT_SHOWUSAGE;
+		else if (ast_unload_resource(argv[x], force)) {
+			ast_cli(fd, "Unable to unload resource %s\n", argv[x]);
+			return RESULT_FAILURE;
+		}
+	}
+	return RESULT_SUCCESS;
+}
+
+#define MODLIST_FORMAT  "%-20s %-40.40s %-10d\n"
+#define MODLIST_FORMAT2 "%-20s %-40.40s %-10s\n"
+
+static pthread_mutex_t climodentrylock = PTHREAD_MUTEX_INITIALIZER;
+static int climodentryfd = -1;
+
+static int modlist_modentry(char *module, char *description, int usecnt)
+{
+	ast_cli(climodentryfd, MODLIST_FORMAT, module, description, usecnt);
+	return 0;
+}
+
+static char modlist_help[] =
+"Usage: show modules\n"
+"       Shows Asterisk modules currently in use, and usage "
+"statistics.\n";
+
+static int handle_modlist(int fd, int argc, char *argv[])
+{
+	if (argc != 2)
+		return RESULT_SHOWUSAGE;
+	pthread_mutex_lock(&climodentrylock);
+	climodentryfd = fd;
+	ast_cli(fd, MODLIST_FORMAT2, "Module", "Description", "Use Count");
+	ast_update_module_list(modlist_modentry);
+	climodentryfd = -1;
+	pthread_mutex_unlock(&climodentrylock);
+	return RESULT_SUCCESS;
+}
+
+static int handle_chanlist(int fd, int argc, char *argv[])
+{
+#define FORMAT_STRING  "%15s  (%-10s %-12s %-4d)  %-12s  %-15s\n"
+#define FORMAT_STRING2 "%15s  (%-10s %-12s %-4s)  %-12s  %-15s\n"
+	struct ast_channel *c=NULL;
+	if (argc != 2)
+		return RESULT_SHOWUSAGE;
+	c = ast_channel_walk(NULL);
+	ast_cli(fd, FORMAT_STRING2, "Channel", "Context", "Extension", "Pri", "Appl.", "Data");
+	while(c) {
+		ast_cli(fd, FORMAT_STRING, c->name, c->context, c->exten, c->priority, 
+		c->appl ? c->appl : "(None)", c->data ? ( strlen(c->data) ? c->data : "(Empty)" ): "(None)");
+		c = ast_channel_walk(c);
+	}
+	return RESULT_SUCCESS;
+}
+
+static char showchan_help[] = 
+"Usage: show channel <channel>\n"
+"       Shows lots of information about the specified channel.\n";
+
+static int handle_showchan(int fd, int argc, char *argv[])
+{
+	struct ast_channel *c=NULL;
+	if (argc != 3)
+		return RESULT_SHOWUSAGE;
+	c = ast_channel_walk(NULL);
+	while(c) {
+		if (!strcasecmp(c->name, argv[2])) {
+			ast_cli(fd, 
+	" -- General --\n"
+	"           Name: %s\n"
+	"           Type: %s\n"
+	"     Translator: %s\n"
+	"         Master: %s\n"
+	"      Caller ID: %s\n"
+	"    DNID Digits: %s\n"
+	"          State: %d\n"
+	"          Rings: %d\n"
+	"         Format: %d\n"
+	"File Descriptor: %d\n"
+	" --   PBX   --\n"
+	"        Context: %s\n"
+	"      Extension: %s\n"
+	"       Priority: %d\n"
+	"    Application: %s\n"
+	"           Data: %s\n"
+	"          Stack: %d\n"
+	"    Blocking in: %s\n",
+	c->name, c->type, (c->trans ? c->trans->name : "(N/A)"),
+	(c->master ? c->master->name : "(N/A)"), (c->callerid ? c->callerid : "(N/A)"),
+	(c->dnid ? c->dnid : "(N/A)" ), c->state, c->rings, c->format,
+	c->fd, c->context, c->exten, c->priority, ( c->appl ? c->appl : "(N/A)" ),
+	( c-> data ? (strlen(c->data) ? c->data : "(Empty)") : "(None)"),
+	c->stack, (c->blocking ? c->blockproc : "(Not Blocking)"));
+	
+		break;
+		}
+		c = ast_channel_walk(c);
+	}
+	if (!c) 
+		ast_cli(fd, "%s is not a known channel\n", argv[2]);
+	return RESULT_SUCCESS;
+}
+
+static char *complete_ch(char *line, char *word, int pos, int state)
+{
+	struct ast_channel *c;
+	int which=0;
+	c = ast_channel_walk(NULL);
+	while(c) {
+		if (++which > state)
+			break;
+		c = ast_channel_walk(c);
+	}
+	return c ? strdup(c->name) : NULL;
+}
+
+static char *complete_fn(char *line, char *word, int pos, int state)
+{
+	char *c;
+	char filename[256];
+	if (pos != 1)
+		return NULL;
+	if (word[0] == '/')
+		strncpy(filename, word, sizeof(filename));
+	else
+		snprintf(filename, sizeof(filename), "%s/%s", AST_MODULE_DIR, word);
+	c = filename_completion_function(filename, state);
+	if (c && word[0] != '/')
+		c += (strlen(AST_MODULE_DIR) + 1);
+	return c ? strdup(c) : c;
+}
+
+static int handle_help(int fd, int argc, char *argv[]);
+
+static struct ast_cli_entry builtins[] = {
+	/* Keep alphabetized */
+	{ { "help", NULL }, handle_help, "Display help list, or specific help on a command", help_help },
+	{ { "load", NULL }, handle_load, "Load a dynamic module by name", load_help, complete_fn },
+	{ { "show", "channel", NULL }, handle_showchan, "Display information on a specific channel", showchan_help, complete_ch },
+	{ { "show", "channels", NULL }, handle_chanlist, "Display information on channels", chanlist_help },
+    { { "show", "modules", NULL }, handle_modlist, "List modules and info", modlist_help },
+	{ { "unload", NULL }, handle_unload, "Unload a dynamic module by name", unload_help, complete_fn },
+	{ { NULL }, NULL, NULL, NULL }
+};
+
+static struct ast_cli_entry *find_cli(char *cmds[], int exact)
+{
+	int x;
+	int y;
+	int match;
+	struct ast_cli_entry *e=NULL;
+	for (x=0;builtins[x].cmda[0];x++) {
+		/* start optimistic */
+		match = 1;
+		for (y=0;match && cmds[y]; y++) {
+			/* If there are no more words in the candidate command, then we're
+			   there.  */
+			if (!builtins[x].cmda[y] && !exact)
+				break;
+			/* If there are no more words in the command (and we're looking for
+			   an exact match) or there is a difference between the two words,
+			   then this is not a match */
+			if (!builtins[x].cmda[y] || strcasecmp(builtins[x].cmda[y], cmds[y]))
+				match = 0;
+		}
+		/* If more words are needed to complete the command then this is not
+		   a candidate (unless we're looking for a really inexact answer  */
+		if ((exact > -1) && builtins[x].cmda[y])
+			match = 0;
+		if (match)
+			return &builtins[x];
+	}
+	for (e=helpers;e;e=e->next) {
+		match = 1;
+		for (y=0;match && cmds[y]; y++) {
+			if (!e->cmda[y] && !exact)
+				break;
+			if (!e->cmda[y] || strcasecmp(e->cmda[y], cmds[y]))
+				match = 0;
+		}
+		if ((exact > -1) && e->cmda[y])
+			match = 0;
+		if (match)
+			break;
+	}
+	return e;
+}
+
+static void join(char *s, int len, char *w[])
+{
+	int x;
+	/* Join words into a string */
+	strcpy(s, "");
+	for (x=0;w[x];x++) {
+		if (x)
+			strncat(s, " ", len - strlen(s));
+		strncat(s, w[x], len - strlen(s));
+	}
+}
+
+static void join2(char *s, int len, char *w[])
+{
+	int x;
+	/* Join words into a string */
+	strcpy(s, "");
+	for (x=0;w[x];x++) {
+		strncat(s, w[x], len - strlen(s));
+	}
+}
+
+static char *find_best(char *argv[])
+{
+	static char cmdline[80];
+	int x;
+	/* See how close we get, then print the  */
+	char *myargv[AST_MAX_CMD_LEN];
+	for (x=0;x<AST_MAX_CMD_LEN;x++)
+		myargv[x]=NULL;
+	for (x=0;argv[x];x++) {
+		myargv[x] = argv[x];
+		if (!find_cli(myargv, -1))
+			break;
+	}
+	join(cmdline, sizeof(cmdline), myargv);
+	return cmdline;
+}
+
+int ast_cli_unregister(struct ast_cli_entry *e)
+{
+	struct ast_cli_entry *cur, *l=NULL;
+	pthread_mutex_lock(&clilock);
+	cur = helpers;
+	while(cur) {
+		if (e == cur) {
+			/* Rewrite */
+			if (l)
+				l->next = e->next;
+			else
+				helpers = e->next;
+			e->next = NULL;
+			break;
+		}
+		l = cur;
+		cur = cur->next;
+	}
+	pthread_mutex_unlock(&clilock);
+	return 0;
+}
+
+int ast_cli_register(struct ast_cli_entry *e)
+{
+	struct ast_cli_entry *cur, *l=NULL;
+	char fulle[80], fulltst[80];
+	static int len;
+	pthread_mutex_lock(&clilock);
+	join2(fulle, sizeof(fulle), e->cmda);
+	if (find_cli(e->cmda, -1)) {
+		ast_log(LOG_WARNING, "Command '%s' already registered (or something close enough)\n", fulle);
+		pthread_mutex_unlock(&clilock);
+		return -1;
+	}
+	cur = helpers;
+	while(cur) {
+		join2(fulltst, sizeof(fulltst), cur->cmda);
+		len = strlen(fulltst);
+		if (strlen(fulle) < len)
+			len = strlen(fulle);
+		if (strncasecmp(fulle, fulltst, len) < 0) {
+			if (l) {
+				e->next = l->next;
+				l->next = e;
+			} else {
+				e->next = helpers;
+				helpers = e;
+			}
+			break;
+		}
+		l = cur;
+		cur = cur->next;
+	}
+	if (!cur) {
+		if (l)
+			l->next = e;
+		else
+			helpers = e;
+		e->next = NULL;
+	}
+	pthread_mutex_unlock(&clilock);
+	return 0;
+}
+
+static int help_workhorse(int fd, char *match[])
+{
+	char fullcmd1[80];
+	char fullcmd2[80];
+	char matchstr[80];
+	char *fullcmd;
+	struct ast_cli_entry *e, *e1, *e2;
+	e1 = builtins;
+	e2 = helpers;
+	if (match)
+		join(matchstr, sizeof(matchstr), match);
+	while(e1->cmda[0] || e2) {
+		if (e2)
+			join(fullcmd2, sizeof(fullcmd2), e2->cmda);
+		if (e1->cmda[0])
+			join(fullcmd1, sizeof(fullcmd1), e1->cmda);
+		if (!e1->cmda[0] || 
+				(e2 && (strcmp(fullcmd2, fullcmd1) < 0))) {
+			/* Use e2 */
+			e = e2;
+			fullcmd = fullcmd2;
+			/* Increment by going to next */
+			e2 = e2->next;
+		} else {
+			/* Use e1 */
+			e = e1;
+			fullcmd = fullcmd1;
+			e1++;
+		}
+		if (match) {
+			if (strncasecmp(matchstr, fullcmd, strlen(matchstr))) {
+				continue;
+			}
+		}
+		ast_cli(fd, "%20.20s   %s\n", fullcmd, e->summary);
+	}
+	return 0;
+}
+
+static int handle_help(int fd, int argc, char *argv[]) {
+	struct ast_cli_entry *e;
+	char fullcmd[80];
+	if ((argc < 1))
+		return RESULT_SHOWUSAGE;
+	if (argc > 1) {
+		e = find_cli(argv + 1, 1);
+		if (e) 
+			ast_cli(fd, e->usage);
+		else {
+			if (find_cli(argv + 1, -1)) {
+				return help_workhorse(fd, argv + 1);
+			} else {
+				join(fullcmd, sizeof(fullcmd), argv+1);
+				ast_cli(fd, "No such command '%s'.\n", fullcmd);
+			}
+		}
+	} else {
+		return help_workhorse(fd, NULL);
+	}
+	return RESULT_SUCCESS;
+}
+
+static char *parse_args(char *s, int *max, char *argv[])
+{
+	char *dup, *cur;
+	int x=0;
+	int quoted=0;
+	int escaped=0;
+	int whitespace=1;
+
+	dup = strdup(s);
+	if (dup) {
+		cur = dup;
+		while(*s) {
+			switch(*s) {
+			case '"':
+				/* If it's escaped, put a literal quote */
+				if (escaped) 
+					goto normal;
+				else 
+					quoted = !quoted;
+				escaped = 0;
+				break;
+			case ' ':
+			case '\t':
+				if (!quoted && !escaped) {
+					/* If we're not quoted, mark this as whitespace, and
+					   end the previous argument */
+					whitespace = 1;
+					*(cur++) = '\0';
+				} else
+					/* Otherwise, just treat it as anything else */ 
+					goto normal;
+				break;
+			case '\\':
+				/* If we're escaped, print a literal, otherwise enable escaping */
+				if (escaped) {
+					goto normal;
+				} else {
+					escaped=1;
+				}
+				break;
+			default:
+normal:
+				if (whitespace) {
+					if (x >= AST_MAX_ARGS -1) {
+						ast_log(LOG_WARNING, "Too many arguments, truncating\n");
+						break;
+					}
+					/* Coming off of whitespace, start the next argument */
+					argv[x++] = cur;
+					whitespace=0;
+				}
+				*(cur++) = *s;
+				escaped=0;
+			}
+			s++;
+		}
+		/* Null terminate */
+		*(cur++) = '\0';
+		argv[x] = NULL;
+		*max = x;
+	}
+	return dup;
+}
+
+char *ast_cli_generator(char *text, char *word, int state)
+{
+	char *argv[AST_MAX_ARGS];
+	struct ast_cli_entry *e, *e1, *e2;
+	int x;
+	int matchnum=0;
+	char *dup, *res;
+	char fullcmd1[80];
+	char fullcmd2[80];
+	char matchstr[80];
+	char *fullcmd;
+
+	if ((dup = parse_args(text, &x, argv))) {
+		join(matchstr, sizeof(matchstr), argv);
+		pthread_mutex_lock(&clilock);
+		e1 = builtins;
+		e2 = helpers;
+		while(e1->cmda[0] || e2) {
+			if (e2)
+				join(fullcmd2, sizeof(fullcmd2), e2->cmda);
+			if (e1->cmda[0])
+				join(fullcmd1, sizeof(fullcmd1), e1->cmda);
+			if (!e1->cmda || 
+					(e2 && (strcmp(fullcmd2, fullcmd1) < 0))) {
+				/* Use e2 */
+				e = e2;
+				fullcmd = fullcmd2;
+				/* Increment by going to next */
+				e2 = e2->next;
+			} else {
+				/* Use e1 */
+				e = e1;
+				fullcmd = fullcmd1;
+				e1++;
+			}
+			if (!strncasecmp(matchstr, fullcmd, strlen(matchstr))) {
+				/* We contain the first part of one or more commands */
+				matchnum++;
+				if (matchnum > state) {
+					/* Now, what we're supposed to return is the next word... */
+					if (strlen(word)) {
+						res = e->cmda[x-1];
+					} else {
+						res = e->cmda[x];
+					}
+					if (res) {
+						pthread_mutex_unlock(&clilock);
+						return res ? strdup(res) : NULL;
+					}
+				}
+			}
+			if (e->generator && !strncasecmp(matchstr, fullcmd, strlen(fullcmd))) {
+				/* We have a command in its entirity within us -- theoretically only one
+				   command can have this occur */
+				fullcmd = e->generator(text, word, (strlen(word) ? (x - 1) : (x)), state);
+				pthread_mutex_unlock(&clilock);
+				return fullcmd;
+			}
+			
+		}
+		pthread_mutex_unlock(&clilock);
+		free(dup);
+	}
+	return NULL;
+}
+
+int ast_cli_command(int fd, char *s)
+{
+	char *argv[AST_MAX_ARGS];
+	struct ast_cli_entry *e;
+	int x;
+	char *dup;
+	x = AST_MAX_ARGS;
+	if ((dup = parse_args(s, &x, argv))) {
+		/* We need at least one entry, or ignore */
+		if (x > 0) {
+			pthread_mutex_lock(&clilock);
+			e = find_cli(argv, 0);
+			if (e) {
+				switch(e->handler(fd, x, argv)) {
+				case RESULT_SHOWUSAGE:
+					ast_cli(fd, e->usage);
+					break;
+				default:
+				}
+			} else 
+				ast_cli(fd, "No such command '%s' (type 'help' for help)\n", find_best(argv));
+			pthread_mutex_unlock(&clilock);
+		}
+		free(dup);
+	} else {
+		ast_log(LOG_WARNING, "Out of memory\n");	
+		return -1;
+	}
+	return 0;
+}

codecs/Makefile

@@ -3,7 +3,7 @@
 # 
 # Makefile for PBX frontends (dynamically loaded)
 #
-# Copyright (C) 1999, Adtran Inc. and Linux Support Services, LLC
+# Copyright (C) 1999, Mark Spencer
 #
 # Mark Spencer <markster@linux-support.net>
 #
@@ -26,17 +26,19 @@ LIBG723=g723.1/libg723.a
 LIBG723B=g723.1b/libg723b.a
 LIBGSM=gsm/lib/libgsm.a
 LIBMP3=mp3/libmp3.a
+LIBLPC10=lpc10/liblpc10.a
 
-CODECS+=$(MODG723) codec_gsm.so #codec_mp3_d.so
+CODECS+=$(MODG723) codec_gsm.so codec_mp3_d.so codec_lpc10.so
 
 all: $(CODECS)
 
 clean:
 	rm -f *.so *.o
-	make -C g723.1 clean
-	make -C g723.1b clean
+	! [ -d g723.1 ] || make -C g723.1 clean
+	! [ -d g723.1b ] || make -C g723.1b clean
 	make -C gsm clean
 	make -C mp3 clean
+	make -C lpc10 clean
 
 $(LIBG723):
 	make -C g723.1 all
@@ -50,11 +52,14 @@ $(LIBG723B):
 $(LIBMP3):
 	make -C mp3 all
 
+$(LIBLPC10):
+	make -C lpc10 all
+
 codec_g723_1.so : codec_g723_1.o $(LIBG723)
 	$(CC) -shared -Xlinker -x -o $@ $< $(LIBG723)
 
 codec_g723_1b.o : codec_g723_1.c
-	$(CC) -c -o $@ $(CFLAGS) -DANNEX_B $<
+	$(CC) -c -o $@ $(CFLAGS) -DANNEX_B -Dsingle $<
 
 codec_g723_1b.so : codec_g723_1b.o $(LIBG723B)
 	$(CC) -shared -Xlinker -x -o $@ $< $(LIBG723B) -lm
@@ -62,6 +67,9 @@ codec_g723_1b.so : codec_g723_1b.o $(LIBG723B)
 codec_gsm.so: codec_gsm.o $(LIBGSM)
 	$(CC) -shared -Xlinker -x -o $@ $< $(LIBGSM)
 
+codec_lpc10.so: codec_lpc10.o $(LIBLPC10)
+	$(CC) -shared -Xlinker -x -o $@ $< $(LIBLPC10) -lm
+
 codec_mp3_d.so: codec_mp3_d.o $(LIBMP3)
 	$(CC) -shared -Xlinker -x -o $@ $< $(LIBMP3)
 

codecs/codec_g723_1.c

@@ -7,7 +7,7 @@
  * it is covered with patents, and in spite of statements to the contrary,
  * the "technology" is extremely expensive to license.
  * 
- * Copyright (C) 1999, Adtran Inc. and Linux Support Services, LLC
+ * Copyright (C) 1999, Mark Spencer
  *
  * Mark Spencer <markster@linux-support.net>
  *
@@ -99,6 +99,8 @@ static struct ast_translator_pvt *g723tolin_new()
 		Init_Decod(&tmp->dec);
 	    Init_Dec_Cng(&tmp->dec);
 		tmp->tail = 0;
+		localusecnt++;
+		ast_update_use_count();
 	}
 	return (struct ast_translator_pvt *)tmp;
 }
@@ -144,6 +146,8 @@ static struct ast_translator_pvt *lintog723_new()
    	   		Init_Vad(&tmp->cod);
         	Init_Cod_Cng(&tmp->cod);
     	 }
+		localusecnt++;
+		ast_update_use_count();
 		tmp->tail = 0;
 	}
 	return (struct ast_translator_pvt *)tmp;
@@ -169,16 +173,14 @@ static struct ast_frame *g723tolin_frameout(struct ast_translator_pvt *pvt)
 	tmp->tail = 0;
 
 #if 0
-	/* Save a sample frame */
-	{ static int samplefr = 0;
-	if (samplefr == 80) {
-		int fd;
-		fd = open("g723.example", O_WRONLY | O_CREAT | O_TRUNC, 0644);
-		write(fd, tmp->f.data, tmp->f.datalen);
-		close(fd);
+	/* Save the frames */
+	{ 
+		static int fd2 = -1;
+		if (fd2 == -1) {
+			fd2 = open("g723.example", O_WRONLY | O_CREAT | O_TRUNC, 0644);
+		}
+		write(fd2, tmp->f.data, tmp->f.datalen);
 	} 		
-	samplefr++;
-	}
 #endif
 	return &tmp->f;	
 }
@@ -196,7 +198,7 @@ static int g723tolin_framein(struct ast_translator_pvt *pvt, struct ast_frame *f
 #ifdef ANNEX_B
 		Decod(&tmp->dec, tmpdata, f->data, 0);
 		for (x=0;x<Frame;x++)
-			(tmp->buf + tmp->tail)[x] = tmpdata[x]; 
+			(tmp->buf + tmp->tail)[x] = (short)(tmpdata[x]); 
 #else
 		Decod(&tmp->dec, tmp->buf + tmp->tail, f->data, 0);
 #endif
@@ -291,6 +293,8 @@ static struct ast_frame *lintog723_frameout(struct ast_translator_pvt *pvt)
 static void g723_destroy(struct ast_translator_pvt *pvt)
 {
 	free(pvt);
+	localusecnt--;
+	ast_update_use_count();
 }
 
 static struct ast_translator g723tolin =

codecs/codec_gsm.c

@@ -6,7 +6,7 @@
  * The GSM code is from TOAST.  Copyright information for that package is available
  * in  the GSM directory.
  * 
- * Copyright (C) 1999, Adtran Inc. and Linux Support Services, LLC
+ * Copyright (C) 1999, Mark Spencer
  *
  * Mark Spencer <markster@linux-support.net>
  *
@@ -65,6 +65,7 @@ static struct ast_translator_pvt *gsm_new()
 			tmp = NULL;
 		}
 		tmp->tail = 0;
+		localusecnt++;
 	}
 	return tmp;
 }
@@ -90,8 +91,8 @@ static struct ast_frame *gsmtolin_sample()
 	f.frametype = AST_FRAME_VOICE;
 	f.subclass = AST_FORMAT_GSM;
 	f.datalen = sizeof(gsm_slin_ex);
-	/* All frames are 30 ms long */
-	f.timelen = 30;
+	/* All frames are 20 ms long */
+	f.timelen = 20;
 	f.mallocd = 0;
 	f.offset = 0;
 	f.src = __PRETTY_FUNCTION__;
@@ -155,8 +156,8 @@ static int lintogsm_framein(struct ast_translator_pvt *tmp, struct ast_frame *f)
 	/* XXX We should look at how old the rest of our stream is, and if it
 	   is too old, then we should overwrite it entirely, otherwise we can
 	   get artifacts of earlier talk that do not belong */
-	if (tmp->tail + f->datalen < sizeof(tmp->buf) / 2) {
-		memcpy(tmp->buf + tmp->tail, f->data, f->datalen);
+	if (tmp->tail + f->datalen/2 < sizeof(tmp->buf) / 2) {
+		memcpy((tmp->buf + tmp->tail), f->data, f->datalen);
 		tmp->tail += f->datalen/2;
 	} else {
 		ast_log(LOG_WARNING, "Out of buffer space\n");
@@ -184,18 +185,16 @@ static struct ast_frame *lintogsm_frameout(struct ast_translator_pvt *tmp)
 	tmp->tail -= 160;
 	/* Move the data at the end of the buffer to the front */
 	if (tmp->tail)
-		memmove(tmp->buf, tmp->buf + 160 * 2, tmp->tail * 2);
+		memmove(tmp->buf, tmp->buf + 160, tmp->tail * 2);
 #if 0
-	/* Save a sample frame */
-	{ static int samplefr = 0;
-	if (samplefr == 0) {
-		int fd;
-		fd = open("gsm.example", O_WRONLY | O_CREAT, 0644);
-		write(fd, tmp->f.data, tmp->f.datalen);
-		close(fd);
+	/* Save the frames */
+	{ 
+		static int fd2 = -1;
+		if (fd2 == -1) {
+			fd2 = open("gsm.example", O_WRONLY | O_CREAT | O_TRUNC, 0644);
+		}
+		write(fd2, tmp->f.data, tmp->f.datalen);
 	} 		
-	samplefr++;
-	}
 #endif
 	return &tmp->f;	
 }
@@ -203,6 +202,7 @@ static struct ast_frame *lintogsm_frameout(struct ast_translator_pvt *tmp)
 static void gsm_destroy_stuff(struct ast_translator_pvt *pvt)
 {
 	free(pvt);
+	localusecnt--;
 }
 
 static struct ast_translator gsmtolin =

codecs/codec_lpc10.c

@@ -0,0 +1,348 @@
+/*
+ * Asterisk -- A telephony toolkit for Linux.
+ *
+ * Translate between signed linear and LPC10 (Linear Predictor Code)
+ *
+ * The lpc10 code is from a library used by nautilus, modified to be a bit
+ * nicer to the compiler.
+ *
+ * See http://www.arl.wustl.edu/~jaf/ 
+ * 
+ * Copyright (C) 1999, Mark Spencer
+ *
+ * Mark Spencer <markster@linux-support.net>
+ *
+ * This program is free software, distributed under the terms of
+ * the GNU General Public License
+ */
+
+
+#include <asterisk/translate.h>
+#include <asterisk/module.h>
+#include <asterisk/logger.h>
+#include <pthread.h>
+#include <fcntl.h>
+#include <stdlib.h>
+#include <unistd.h>
+#include <netinet/in.h>
+#include <string.h>
+#include <stdio.h>
+
+#include "lpc10/lpc10.h"
+
+/* Sample frame data */
+#include "slin_lpc10_ex.h"
+#include "lpc10_slin_ex.h"
+
+/* We use a very strange format here...  I have no idea why...  The frames are 180
+   samples long, which isn't even an even number of milliseconds...  Not only that
+   but we hvae to waste two bits of each frame to keep them ending on a byte boundary
+   because the frames are 54 bits long */
+
+#define LPC10_BYTES_IN_COMPRESSED_FRAME (LPC10_BITS_IN_COMPRESSED_FRAME + 7)/8
+
+static pthread_mutex_t localuser_lock = PTHREAD_MUTEX_INITIALIZER;
+static int localusecnt=0;
+
+static char *tdesc = "LPC10 2.4kbps (signed linear) Voice Coder";
+
+struct ast_translator_pvt {
+	union {
+		struct lpc10_encoder_state *enc;
+		struct lpc10_decoder_state *dec;
+	} lpc10;
+	struct ast_frame f;
+	/* Space to build offset */
+	char offset[AST_FRIENDLY_OFFSET];
+	/* Buffer for our outgoing frame */
+	short outbuf[LPC10_SAMPLES_PER_FRAME];
+	/* Enough to store a full second */
+	short buf[8000];
+	int tail;
+	int longer;
+};
+
+#define lpc10_coder_pvt ast_translator_pvt
+
+static struct ast_translator_pvt *lpc10_enc_new()
+{
+	struct lpc10_coder_pvt *tmp;
+	tmp = malloc(sizeof(struct lpc10_coder_pvt));
+	if (tmp) {
+		if (!(tmp->lpc10.enc = create_lpc10_encoder_state())) {
+			free(tmp);
+			tmp = NULL;
+		}
+		tmp->tail = 0;
+		tmp->longer = 0;
+		localusecnt++;
+	}
+	return tmp;
+}
+
+static struct ast_translator_pvt *lpc10_dec_new()
+{
+	struct lpc10_coder_pvt *tmp;
+	tmp = malloc(sizeof(struct lpc10_coder_pvt));
+	if (tmp) {
+		if (!(tmp->lpc10.dec = create_lpc10_decoder_state())) {
+			free(tmp);
+			tmp = NULL;
+		}
+		tmp->tail = 0;
+		tmp->longer = 0;
+		localusecnt++;
+	}
+	return tmp;
+}
+static struct ast_frame *lintolpc10_sample()
+{
+	static struct ast_frame f;
+	static int longer = 0;
+	f.frametype = AST_FRAME_VOICE;
+	f.subclass = AST_FORMAT_SLINEAR;
+	f.datalen = sizeof(slin_lpc10_ex);
+	/* Assume 8000 Hz */
+	f.timelen = LPC10_SAMPLES_PER_FRAME/8;
+	f.timelen += longer;
+	longer = 1- longer;
+	f.mallocd = 0;
+	f.offset = 0;
+	f.src = __PRETTY_FUNCTION__;
+	f.data = slin_lpc10_ex;
+	return &f;
+}
+
+static struct ast_frame *lpc10tolin_sample()
+{
+	static struct ast_frame f;
+	f.frametype = AST_FRAME_VOICE;
+	f.subclass = AST_FORMAT_LPC10;
+	f.datalen = sizeof(lpc10_slin_ex);
+	/* All frames are 22 ms long (maybe a little more -- why did he choose
+	   LPC10_SAMPLES_PER_FRAME sample frames anyway?? */
+	f.timelen = LPC10_SAMPLES_PER_FRAME/8;
+	f.mallocd = 0;
+	f.offset = 0;
+	f.src = __PRETTY_FUNCTION__;
+	f.data = lpc10_slin_ex;
+	return &f;
+}
+
+static struct ast_frame *lpc10tolin_frameout(struct ast_translator_pvt *tmp)
+{
+	if (!tmp->tail)
+		return NULL;
+	/* Signed linear is no particular frame size, so just send whatever
+	   we have in the buffer in one lump sum */
+	tmp->f.frametype = AST_FRAME_VOICE;
+	tmp->f.subclass = AST_FORMAT_SLINEAR;
+	tmp->f.datalen = tmp->tail * 2;
+	/* Assume 8000 Hz */
+	tmp->f.timelen = tmp->tail / 8;
+	tmp->f.mallocd = 0;
+	tmp->f.offset = AST_FRIENDLY_OFFSET;
+	tmp->f.src = __PRETTY_FUNCTION__;
+	tmp->f.data = tmp->buf;
+	/* Reset tail pointer */
+	tmp->tail = 0;
+
+#if 0
+	/* Save a sample frame */
+	{ static int samplefr = 0;
+	if (samplefr == 80) {
+		int fd;
+		fd = open("lpc10.example", O_WRONLY | O_CREAT, 0644);
+		write(fd, tmp->f.data, tmp->f.datalen);
+		close(fd);
+	} 		
+	samplefr++;
+	}
+#endif
+	return &tmp->f;	
+}
+
+static void extract_bits(INT32 *bits, unsigned char *c)
+{
+	int x;
+	for (x=0;x<LPC10_BITS_IN_COMPRESSED_FRAME;x++) {
+		if (*c & (0x80 >> (x & 7)))
+			bits[x] = 1;
+		else
+			bits[x] = 0;
+		if ((x & 7) == 7)
+			c++;
+	}
+}
+
+static void build_bits(unsigned char *c, INT32 *bits)
+{
+	unsigned char mask=0x80;
+	int x;
+	*c = 0;
+	for (x=0;x<LPC10_BITS_IN_COMPRESSED_FRAME;x++) {
+		if (bits[x])
+			*c |= mask;
+		mask = mask >> 1;
+		if ((x % 8)==7) {
+			c++;
+			*c = 0;
+			mask = 0x80;
+		}
+	}
+}
+
+static int lpc10tolin_framein(struct ast_translator_pvt *tmp, struct ast_frame *f)
+{
+	/* Assuming there's space left, decode into the current buffer at
+	   the tail location */
+	int x;
+	float tmpbuf[LPC10_SAMPLES_PER_FRAME];
+	short *sd;
+	INT32 bits[LPC10_BITS_IN_COMPRESSED_FRAME];
+	if (tmp->tail + LPC10_SAMPLES_PER_FRAME < sizeof(tmp->buf)/2) {
+		sd = tmp->buf + tmp->tail;
+		extract_bits(bits, f->data);
+		if (lpc10_decode(bits, tmpbuf, tmp->lpc10.dec)) {
+			ast_log(LOG_WARNING, "Invalid lpc10 data\n");
+			return -1;
+		}
+		for (x=0;x<LPC10_SAMPLES_PER_FRAME;x++) {
+			/* Convert to a real between -1.0 and 1.0 */
+			sd[x] = 32768.0 * tmpbuf[x];
+		}
+		
+		tmp->tail+=LPC10_SAMPLES_PER_FRAME;
+	} else {
+		ast_log(LOG_WARNING, "Out of buffer space\n");
+		return -1;
+	}
+	return 0;
+}
+
+static int lintolpc10_framein(struct ast_translator_pvt *tmp, struct ast_frame *f)
+{
+	/* Just add the frames to our stream */
+	/* XXX We should look at how old the rest of our stream is, and if it
+	   is too old, then we should overwrite it entirely, otherwise we can
+	   get artifacts of earlier talk that do not belong */
+	if (tmp->tail + f->datalen < sizeof(tmp->buf) / 2) {
+		memcpy((tmp->buf + tmp->tail), f->data, f->datalen);
+		tmp->tail += f->datalen/2;
+	} else {
+		ast_log(LOG_WARNING, "Out of buffer space\n");
+		return -1;
+	}
+	return 0;
+}
+
+static struct ast_frame *lintolpc10_frameout(struct ast_translator_pvt *tmp)
+{
+	int x;
+	float tmpbuf[LPC10_SAMPLES_PER_FRAME];
+	INT32 bits[LPC10_BITS_IN_COMPRESSED_FRAME];
+	/* We can't work on anything less than a frame in size */
+	if (tmp->tail < LPC10_SAMPLES_PER_FRAME)
+		return NULL;
+	/* Encode a frame of data */
+	for (x=0;x<LPC10_SAMPLES_PER_FRAME;x++) {
+		tmpbuf[x] = (float)tmp->buf[x] / 32768.0;
+	}
+	lpc10_encode(tmpbuf, bits, tmp->lpc10.enc);
+	build_bits((unsigned char *)tmp->outbuf, bits);
+	tmp->f.frametype = AST_FRAME_VOICE;
+	tmp->f.subclass = AST_FORMAT_LPC10;
+	tmp->f.datalen = LPC10_BYTES_IN_COMPRESSED_FRAME;
+	tmp->f.timelen = 22;
+	/* We alternate between 22 and 23 ms to simulate 22.5 ms */
+	tmp->f.timelen += tmp->longer;
+	/* Use one of the two left over bits to record if this is a 22 or 23 ms frame...
+	   important for IAX use */
+	tmp->longer = 1 - tmp->longer;
+	tmp->f.mallocd = 0;
+	tmp->f.offset = AST_FRIENDLY_OFFSET;
+	tmp->f.src = __PRETTY_FUNCTION__;
+	tmp->f.data = tmp->outbuf;
+	((char *)(tmp->f.data))[LPC10_BYTES_IN_COMPRESSED_FRAME - 1] |= tmp->longer;
+	tmp->tail -= LPC10_SAMPLES_PER_FRAME;
+	/* Move the data at the end of the buffer to the front */
+	if (tmp->tail)
+		memmove(tmp->buf, tmp->buf + LPC10_SAMPLES_PER_FRAME, tmp->tail * 2);
+#if 0
+	/* Save a sample frame */
+	{ static int samplefr = 0;
+	if (samplefr == 0) {
+		int fd;
+		fd = open("lpc10.example", O_WRONLY | O_CREAT, 0644);
+		write(fd, tmp->f.data, tmp->f.datalen);
+		close(fd);
+	} 		
+	samplefr++;
+	}
+#endif
+	return &tmp->f;	
+}
+
+static void lpc10_destroy(struct ast_translator_pvt *pvt)
+{
+	/* Enc and DEC are both just allocated, so they can be freed */
+	free(pvt->lpc10.enc);
+	free(pvt);
+	localusecnt--;
+}
+
+static struct ast_translator lpc10tolin =
+	{ "lpc10tolin", 
+	   AST_FORMAT_LPC10, AST_FORMAT_SLINEAR,
+	   lpc10_dec_new,
+	   lpc10tolin_framein,
+	   lpc10tolin_frameout,
+	   lpc10_destroy,
+	   lpc10tolin_sample
+	   };
+
+static struct ast_translator lintolpc10 =
+	{ "lintolpc10", 
+	   AST_FORMAT_SLINEAR, AST_FORMAT_LPC10,
+	   lpc10_enc_new,
+	   lintolpc10_framein,
+	   lintolpc10_frameout,
+	   lpc10_destroy,
+	   lintolpc10_sample
+	   };
+
+int unload_module(void)
+{
+	int res;
+	pthread_mutex_lock(&localuser_lock);
+	res = ast_unregister_translator(&lintolpc10);
+	if (!res)
+		res = ast_unregister_translator(&lpc10tolin);
+	if (localusecnt)
+		res = -1;
+	pthread_mutex_unlock(&localuser_lock);
+	return res;
+}
+
+int load_module(void)
+{
+	int res;
+	res=ast_register_translator(&lpc10tolin);
+	if (!res) 
+		res=ast_register_translator(&lintolpc10);
+	else
+		ast_unregister_translator(&lpc10tolin);
+	return res;
+}
+
+char *description(void)
+{
+	return tdesc;
+}
+
+int usecount(void)
+{
+	int res;
+	STANDARD_USECOUNT(res);
+	return res;
+}

codecs/codec_mp3_d.c

@@ -0,0 +1,323 @@
+/*
+ * Asterisk -- A telephony toolkit for Linux.
+ *
+ * MP3 Decoder
+ *
+ * The MP3 code is from freeamp, which in turn is from xingmp3's release
+ * which I can't seem to find anywhere
+ * 
+ * Copyright (C) 1999, Mark Spencer
+ *
+ * Mark Spencer <markster@linux-support.net>
+ *
+ * This program is free software, distributed under the terms of
+ * the GNU General Public License
+ */
+
+#include <asterisk/translate.h>
+#include <asterisk/module.h>
+#include <asterisk/logger.h>
+#include <pthread.h>
+#include <fcntl.h>
+#include <errno.h>
+#include <stdlib.h>
+#include <unistd.h>
+#include <netinet/in.h>
+#include <string.h>
+#include <stdio.h>
+
+#include "mp3/include/L3.h"
+#include "mp3/include/mhead.h"
+
+#include "mp3anal.h"
+
+/* Sample frame data */
+#include "mp3_slin_ex.h"
+
+#define MAX_OUT_FRAME 320
+
+#define MAX_FRAME_SIZE 1441
+#define MAX_OUTPUT_LEN 2304
+
+static pthread_mutex_t localuser_lock = PTHREAD_MUTEX_INITIALIZER;
+static int localusecnt=0;
+
+static char *tdesc = "MP3/PCM16 (signed linear) Translator (Decoder only)";
+
+struct ast_translator_pvt {
+	MPEG m;
+	MPEG_HEAD head;
+	DEC_INFO info;
+	struct ast_frame f;
+	/* Space to build offset */
+	char offset[AST_FRIENDLY_OFFSET];
+	/* Mini buffer */
+	char outbuf[MAX_OUT_FRAME];
+	/* Enough to store a full second */
+	short buf[32000];
+	/* Tail of signed linear stuff */
+	int tail;
+	/* Current bitrate */
+	int bitrate;
+	/* XXX What's forward? XXX */
+	int forward;
+	/* Have we called head info yet? */
+	int init;
+	int copy;
+};
+
+#define mp3_coder_pvt ast_translator_pvt
+
+static struct ast_translator_pvt *mp3_new()
+{
+	struct mp3_coder_pvt *tmp;
+	tmp = malloc(sizeof(struct mp3_coder_pvt));
+	if (tmp) {
+		tmp->init = 0;
+		tmp->tail = 0;
+		tmp->copy = -1;
+		mpeg_init(&tmp->m);
+	}
+	return tmp;
+}
+
+static struct ast_frame *mp3tolin_sample()
+{
+	static struct ast_frame f;
+	int size;
+	if (mp3_badheader(mp3_slin_ex)) {
+		ast_log(LOG_WARNING, "Bad MP3 sample??\n");
+		return NULL;
+	}
+	size = mp3_framelen(mp3_slin_ex);
+	if (size < 1) {
+		ast_log(LOG_WARNING, "Failed to size??\n");
+		return NULL;
+	}
+	f.frametype = AST_FRAME_VOICE;
+	f.subclass = AST_FORMAT_MP3;
+	f.data = mp3_slin_ex;
+	f.datalen = sizeof(mp3_slin_ex);
+	/* Dunno how long an mp3 frame is -- kinda irrelevant anyway */
+	f.timelen = 30;
+	f.mallocd = 0;
+	f.offset = 0;
+	f.src = __PRETTY_FUNCTION__;
+	return &f;
+}
+
+static struct ast_frame *mp3tolin_frameout(struct ast_translator_pvt *tmp)
+{
+	int sent;
+	if (!tmp->tail)
+		return NULL;
+	sent = tmp->tail;
+	if (sent > MAX_OUT_FRAME/2)
+		sent = MAX_OUT_FRAME/2;
+	/* Signed linear is no particular frame size, so just send whatever
+	   we have in the buffer in one lump sum */
+	tmp->f.frametype = AST_FRAME_VOICE;
+	tmp->f.subclass = AST_FORMAT_SLINEAR;
+	tmp->f.datalen = sent * 2;
+	/* Assume 8000 Hz */
+	tmp->f.timelen = sent / 8;
+	tmp->f.mallocd = 0;
+	tmp->f.offset = AST_FRIENDLY_OFFSET;
+	tmp->f.src = __PRETTY_FUNCTION__;
+	memcpy(tmp->outbuf, tmp->buf, tmp->tail * 2);
+	tmp->f.data = tmp->outbuf;
+	/* Reset tail pointer */
+	tmp->tail -= sent;
+	if (tmp->tail) 
+		memmove(tmp->buf, tmp->buf + sent, tmp->tail * 2);
+
+#if 0
+	/* Save a sample frame */
+	{ static int samplefr = 0;
+	if (samplefr == 80) {
+		int fd;
+		fd = open("mp3.example", O_WRONLY | O_CREAT, 0644);
+		write(fd, tmp->f.data, tmp->f.datalen);
+		close(fd);
+	} 		
+	samplefr++;
+	}
+#endif
+	return &tmp->f;	
+}
+
+static int mp3_init(struct ast_translator_pvt *tmp, int len)
+{	
+	if (!audio_decode_init(&tmp->m, &tmp->head, len,0,0,1 /* Convert to mono */,24000)) {
+		ast_log(LOG_WARNING, "audio_decode_init() failed\n");
+		return -1;
+	}
+	audio_decode_info(&tmp->m, &tmp->info);
+#if 0
+	ast_verbose(
+"Channels: %d\nOutValues: %d\nSample Rate: %d\nBits: %d\nFramebytes: %d\nType: %d\n",
+	tmp->info.channels, tmp->info.outvalues, tmp->info.samprate, tmp->info.bits,tmp->info.framebytes,tmp->info.type);
+#endif
+	return 0;
+}
+
+#ifndef MIN
+#define MIN(a,b) (((a) < (b)) ? (a) : (b))
+#endif
+
+#if 1
+static int add_to_buf(short *dst, int maxdst, short *src, int srclen, int samprate)
+{
+	float inc, cur, sum=0;
+	int cnt=0, pos, ptr, lastpos = -1;
+	/* Resample source to destination converting from its sampling rate to 8000 Hz */
+	if (samprate == 8000) {
+		/* Quickly, all we have to do is copy */
+		memcpy(dst, src, 2 * MIN(maxdst, srclen));
+		return MIN(maxdst, srclen);
+	}
+	if (samprate < 8000) {
+		ast_log(LOG_WARNING, "Don't know how to resample a source less than 8000 Hz!\n");
+		/* XXX Wrong thing to do XXX */
+		memcpy(dst, src, 2 * MIN(maxdst, srclen));
+		return MIN(maxdst, srclen);
+	}
+	/* Ugh, we actually *have* to resample */
+	inc = 8000.0 / (float)samprate;
+	cur = 0;
+	ptr = 0;
+	pos = 0;
+#if 0
+	ast_verbose("Incrementing by %f, in = %d bytes, out = %d bytes\n", inc, srclen, maxdst);
+#endif
+	while((pos < maxdst) && (ptr < srclen)) {
+		if (pos != lastpos) {
+			if (lastpos > -1) {
+				sum = sum / (float)cnt;
+				dst[pos - 1] = (int) sum;
+#if 0
+				ast_verbose("dst[%d] = %d\n", pos - 1, dst[pos - 1]);
+#endif
+			}
+			/* Each time we have a first pass */
+			sum = 0;
+			cnt = 0;
+		} else {
+			sum += src[ptr];
+		}
+		ptr++;
+		cur += inc;
+		cnt++;
+		lastpos = pos;
+		pos = (int)cur;
+	}
+	return pos;
+}
+#endif
+
+static int mp3tolin_framein(struct ast_translator_pvt *tmp, struct ast_frame *f)
+{
+	/* Assuming there's space left, decode into the current buffer at
+	   the tail location */
+	int framelen;
+	short tmpbuf[8000];
+	IN_OUT x;
+#if 0
+	if (tmp->copy < 0) {
+		tmp->copy = open("sample.out", O_WRONLY | O_CREAT | O_TRUNC, 0700);
+	}
+	if (tmp->copy > -1)
+		write(tmp->copy, f->data, f->datalen);
+#endif
+	/* Check if it's a valid frame */
+	if (mp3_badheader((unsigned char *)f->data)) {
+		ast_log(LOG_WARNING, "Invalid MP3 header\n");
+		return -1;
+	}
+	if ((framelen = mp3_framelen((unsigned char *)f->data) != f->datalen)) {
+		ast_log(LOG_WARNING, "Calculated length %d does not match real length %d\n", framelen, f->datalen);
+		return -1;
+	}
+	/* Start by putting this in the mp3 buffer */
+	if((framelen = head_info3(f->data, 
+			f->datalen, &tmp->head, &tmp->bitrate, &tmp->forward)) > 0) {
+		if (!tmp->init) {
+			if (mp3_init(tmp, framelen))
+				return -1;
+			else
+				tmp->init++;
+		}
+		if (tmp->tail + MAX_OUTPUT_LEN/2  < sizeof(tmp->buf)/2) {	
+			x = audio_decode(&tmp->m, f->data, tmpbuf);
+			audio_decode_info(&tmp->m, &tmp->info);
+			if (!x.in_bytes) {
+				ast_log(LOG_WARNING, "Invalid MP3 data\n");
+			} else {
+#if 1
+				/* Resample to 8000 Hz */
+				tmp->tail += add_to_buf(tmp->buf + tmp->tail, 
+			           sizeof(tmp->buf) / 2 - tmp->tail, 
+					   tmpbuf,
+					   x.out_bytes/2,
+					   tmp->info.samprate);
+#else
+				memcpy(tmp->buf + tmp->tail, tmpbuf, x.out_bytes);
+				/* Signed linear output */
+				tmp->tail+=x.out_bytes/2;
+#endif
+			}
+		} else {
+			ast_log(LOG_WARNING, "Out of buffer space\n");
+			return -1;
+		}
+	} else {
+		ast_log(LOG_WARNING, "Not a valid MP3 frame\n");
+	}
+	return 0;
+}
+
+static void mp3_destroy_stuff(struct ast_translator_pvt *pvt)
+{
+	close(pvt->copy);
+	free(pvt);
+}
+
+static struct ast_translator mp3tolin =
+	{ "mp3tolin", 
+	   AST_FORMAT_MP3, AST_FORMAT_SLINEAR,
+	   mp3_new,
+	   mp3tolin_framein,
+	   mp3tolin_frameout,
+	   mp3_destroy_stuff,
+	   mp3tolin_sample
+	   };
+
+int unload_module(void)
+{
+	int res;
+	pthread_mutex_lock(&localuser_lock);
+	res = ast_unregister_translator(&mp3tolin);
+	if (localusecnt)
+		res = -1;
+	pthread_mutex_unlock(&localuser_lock);
+	return res;
+}
+
+int load_module(void)
+{
+	int res;
+	res=ast_register_translator(&mp3tolin);
+	return res;
+}
+
+char *description(void)
+{
+	return tdesc;
+}
+
+int usecount(void)
+{
+	int res;
+	STANDARD_USECOUNT(res);
+	return res;
+}

codecs/g723_slin_ex.h

@@ -3,7 +3,7 @@
   *
   * Source: g723.example
   *
-  * Copyright (C) 1999, Mark Spencer and Linux Support Services
+  * Copyright (C) 1999, Mark Spencer
   *
   * Distributed under the terms of the GNU General Public License
   *

codecs/gsm_slin_ex.h

@@ -3,7 +3,7 @@
   *
   * Source: gsm.example
   *
-  * Copyright (C) 1999, Mark Spencer and Linux Support Services
+  * Copyright (C) 1999, Mark Spencer
   *
   * Distributed under the terms of the GNU General Public License
   *

codecs/lpc10/Makefile

@@ -0,0 +1,70 @@
+# 
+# Makefile for LPC-10 speech coder library (unix)
+# 
+
+# default C compiler
+CC= gcc
+
+# 
+# These definitions for CFLAGS and LIB_TARGET_DIR are used when one
+# runs make in the lpc10 directory, without environment variables that
+# override them.  When make is run in this directory from a makefile
+# for an application that uses the LPC10 coder, there are environment
+# variables set for CFLAGS and LIB_TARGET_DIR that override these
+# definitions.
+# 
+
+LIB_TARGET_DIR = .
+
+# 
+# -I$(LIB_TARGET_DIR) option needed so that #include "machine.h"
+# directives can find the machine.h file.
+# 
+
+WARNINGS = -Wall -Wno-comment -Wno-error
+CFLAGS = -O3 -I$(LIB_TARGET_DIR) $(WARNINGS)
+
+LIB = $(LIB_TARGET_DIR)/liblpc10.a
+
+.PHONY: all clean
+
+all:	$(LIB)
+	ranlib $(LIB)
+
+$(LIB):		$(LIB)(f2clib.o)	\
+		$(LIB)(analys.o)	\
+		$(LIB)(bsynz.o)		\
+		$(LIB)(chanwr.o)	\
+		$(LIB)(dcbias.o)	\
+		$(LIB)(decode.o)	\
+		$(LIB)(deemp.o)		\
+		$(LIB)(difmag.o)	\
+		$(LIB)(dyptrk.o)	\
+		$(LIB)(encode.o)	\
+		$(LIB)(energy.o)	\
+		$(LIB)(ham84.o)		\
+		$(LIB)(hp100.o)		\
+		$(LIB)(invert.o)	\
+		$(LIB)(irc2pc.o)	\
+		$(LIB)(ivfilt.o)	\
+		$(LIB)(lpcdec.o)	\
+		$(LIB)(lpcenc.o)	\
+		$(LIB)(lpcini.o)	\
+		$(LIB)(lpfilt.o)	\
+		$(LIB)(median.o)	\
+		$(LIB)(mload.o)		\
+		$(LIB)(onset.o)		\
+		$(LIB)(pitsyn.o)	\
+		$(LIB)(placea.o)	\
+		$(LIB)(placev.o)	\
+		$(LIB)(preemp.o)	\
+		$(LIB)(prepro.o)	\
+		$(LIB)(random.o)	\
+		$(LIB)(rcchk.o)		\
+		$(LIB)(synths.o)	\
+		$(LIB)(tbdm.o)		\
+		$(LIB)(voicin.o)	\
+		$(LIB)(vparms.o)
+
+clean:
+	-rm -f *.o $(LIB)

codecs/lpc10/README

@@ -0,0 +1,89 @@
+Tue Aug 20 16:19:51 CDT 1996
+Andy Fingerhut (jaf@arl.wustl.edu)
+
+In release 1.4, there are quite a few hand modifications to the C code
+that was automatically created from the Fortran code with f2c.  They
+are all summarized in change log comments at the beginning of the
+changed files.  All of the original files from f2c were checked in to
+RCS before modification, so it is possible to see exactly what changes
+were made, for the extremely curious.  That precaution was also for my
+benefit, in case I ever recompile the Fortran sources, and want to
+make similar changes to that new C source code.
+
+Below is the README file for this directory included with the 1.3
+release of the LPC-10 package.  A few parts of it are a little out of
+date, but it is correct for the most part.
+
+
+Sun Jul  7 15:30:31 CDT 1996
+Andy Fingerhut (jaf@arl.wustl.edu)
+
+To create the LPC-10 library, copy the appropriate makefile to the
+proper name for easy use, e.g., for Unix, copy makefile.unx to the
+file "Makefile".  The file makefile.dos has been used with some
+version of the 'nmake' utility that comes with the Microsoft C
+compiler (the same one used for Nautilus v1.5a, which I believe
+specifies Microsoft C version 7.0 or later).
+
+Then edit the file lpc10.h in the directory above.  It should already
+be set up to work properly on any Unix compiler for which "int" is 32
+bits and "short" is 16 bits, and under the Microsoft C compiler
+configured so that "long" is 32 bits and "int" is 16 bits.  There must
+be a typedef for the two types INT32 and INT16 in that file.  You
+should choose types that compile to those sizes using your compiler,
+because there are places in the LPC-10 code that expect INT16's to
+have exactly 16 bits (at least, I *think* they must be no larger), and
+INT32's to have exactly 32 bits.
+
+
+A few notes on how these files were created
+-------------------------------------------
+
+(This section is mostly for my benefit, so I can remember what I did.
+You don't need to read it if you just want to use this package.  It
+might be useful to read it if you change the Fortran sources and want
+to recreate a usable library of C sources.  -- Andy)
+
+These C sources were created automatically from the Fortran sources
+using f2c, for the most part.  Listed below are the extra
+modifications that were made after this automatic conversion.  Many of
+them were made so that it was not necessary to install f2c in order to
+use this LPC-10 coder.
+
+1.
+
+Put all of those files that were necessary for only the coder, rather
+than an application that uses the coder, into this subdirectory called
+lpc10.
+
+2.
+
+Copied f2c.h from the f2c distribution into this subdirectory.  Some
+modifications were made to the "typedef" statements in this file, to
+explicitly indicate the sizes (in bits) that different integer types
+should be.  The types INT32 and INT16 must be defined in a file called
+lpc10.h in the directory above.  Created the file f2clib.c, containing
+only the functions pow_ii(), r_sign(), and i_nint() from the f2c
+library.
+
+3.
+
+The f2c output originally had a file called contrl_com.c, that defined
+a small structure containing a few variables that were used in many
+different functions of the LPC10 code.  Every file containing
+functions that used it defined it as "extern", while contrl_com.c
+actually allocated storage for the structure.  Bill Dorsey, one of the
+lead developers of Nautilus, said that the Microsoft C compiler had
+problems either compiling this file, or linking it with all of the
+other compiled files, so he just eliminated that file and removed the
+"extern" keyword from the one of the files that declared it that way.
+The file chosen (arbitrarily) was analys.c.
+
+4.
+
+Copied the makefiles for Unix and Microsoft C from the Nautilus v1.5a
+distribution into the lpc10 directory.  Modified them to take out
+references to Nautilus.  These makefiles don't create an executable,
+but a library of compiled functions called liblpc10.a (Unix) or
+LPC10.LIB (DOS).  This library can be used when linking an executable
+that calls the functions lpcini_(), lpcenc_(), and lpcdec_().

codecs/lpc10/analys.c

@@ -0,0 +1,617 @@
+/*
+
+$Log$
+Revision 1.4  2000/01/05 08:20:39  markster
+Version 0.1.5 from FTP
+
+Revision 1.2  2000/01/05 08:20:39  markster
+Some OSS fixes and a few lpc changes to make it actually work
+
+ * Revision 1.2  1996/08/20  20:16:01  jaf
+ * Removed all static local variables that were SAVE'd in the Fortran
+ * code, and put them in struct lpc10_encoder_state that is passed as an
+ * argument.
+ *
+ * Removed init function, since all initialization is now done in
+ * init_lpc10_encoder_state().
+ *
+ * Revision 1.1  1996/08/19  22:29:08  jaf
+ * Initial revision
+ *
+
+*/
+
+#ifdef P_R_O_T_O_T_Y_P_E_S
+extern int analys_(real *speech, integer *voice, integer *pitch, real *rms, real *rc, struct lpc10_encoder_state *st);
+/* comlen contrl_ 12 */
+/*:ref: preemp_ 14 5 6 6 4 6 6 */
+/*:ref: onset_ 14 7 6 4 4 4 4 4 4 */
+/*:ref: placev_ 14 11 4 4 4 4 4 4 4 4 4 4 4 */
+/*:ref: lpfilt_ 14 4 6 6 4 4 */
+/*:ref: ivfilt_ 14 5 6 6 4 4 6 */
+/*:ref: tbdm_ 14 8 6 4 4 4 6 4 4 4 */
+/*:ref: voicin_ 14 12 4 6 6 4 4 6 6 4 6 4 4 4 */
+/*:ref: dyptrk_ 14 6 6 4 4 4 4 4 */
+/*:ref: placea_ 14 9 4 4 4 4 4 4 4 4 4 */
+/*:ref: dcbias_ 14 3 4 6 6 */
+/*:ref: energy_ 14 3 4 6 6 */
+/*:ref: mload_ 14 6 4 4 4 6 6 6 */
+/*:ref: invert_ 14 4 4 6 6 6 */
+/*:ref: rcchk_ 14 3 4 6 6 */
+/*:ref: initonset_ 14 0 */
+/*:ref: initvoicin_ 14 0 */
+/*:ref: initdyptrk_ 14 0 */
+/* Rerunning f2c -P may change prototypes or declarations. */
+#endif
+
+/*  -- translated by f2c (version 19951025).
+   You must link the resulting object file with the libraries:
+	-lf2c -lm   (in that order)
+*/
+
+#include "f2c.h"
+
+/* Common Block Declarations */
+
+extern struct {
+    integer order, lframe;
+    logical corrp;
+} contrl_;
+
+#define contrl_1 contrl_
+
+/* Table of constant values */
+
+static integer c__10 = 10;
+static integer c__181 = 181;
+static integer c__720 = 720;
+static integer c__3 = 3;
+static integer c__90 = 90;
+static integer c__156 = 156;
+static integer c__307 = 307;
+static integer c__462 = 462;
+static integer c__312 = 312;
+static integer c__60 = 60;
+static integer c__1 = 1;
+
+/* ****************************************************************** */
+
+/* 	ANALYS Version 55 */
+
+/* $Log$
+ * Revision 1.4  2000/01/05 08:20:39  markster
+ * Version 0.1.5 from FTP
+ *
+/* Revision 1.2  2000/01/05 08:20:39  markster
+/* Some OSS fixes and a few lpc changes to make it actually work
+/*
+ * Revision 1.2  1996/08/20  20:16:01  jaf
+ * Removed all static local variables that were SAVE'd in the Fortran
+ * code, and put them in struct lpc10_encoder_state that is passed as an
+ * argument.
+ *
+ * Removed init function, since all initialization is now done in
+ * init_lpc10_encoder_state().
+ *
+ * Revision 1.1  1996/08/19  22:29:08  jaf
+ * Initial revision
+ * */
+/* Revision 1.9  1996/05/23  19:41:07  jaf */
+/* Commented out some unnecessary lines that were reading uninitialized */
+/* values. */
+
+/* Revision 1.8  1996/03/27  23:57:55  jaf */
+/* Added some comments about which indices of the local buffers INBUF, */
+/* LPBUF, etc., get read or modified by some of the subroutine calls.  I */
+/* just did this while trying to figure out the discrepancy between the */
+/* embedded code compiled with all local variables implicitly saved, and */
+/* without. */
+
+/* I added some debugging write statements in hopes of finding a problem. */
+/* None of them ever printed anything while running with the long input */
+/* speech file dam9.spd provided in the distribution. */
+
+/* Revision 1.7  1996/03/27  18:06:20  jaf */
+/* Commented out access to MAXOSP, which is just a debugging variable */
+/* that was defined in the COMMON block CONTRL in contrl.fh. */
+
+/* Revision 1.6  1996/03/26  19:31:33  jaf */
+/* Commented out trace statements. */
+
+/* Revision 1.5  1996/03/21  15:19:35  jaf */
+/* Added comments for ENTRY PITDEC. */
+
+/* Revision 1.4  1996/03/19  20:54:27  jaf */
+/* Added a line to INITANALYS.  See comments there. */
+
+/* Revision 1.3  1996/03/19  20:52:49  jaf */
+/* Rearranged the order of the local variables quite a bit, to separate */
+/* them into groups of "constants", "locals that don't need to be saved */
+/* from one call to the next", and "local that do need to be saved from */
+/* one call to the next". */
+
+/* Several locals in the last set should have been given initial values, */
+/* but weren't.  I gave them all initial values of 0. */
+
+/* Added a separate ENTRY INITANALYS that initializes all local state */
+/* that should be, and also calls the corresponding entries of the */
+/* subroutines called by ANALYS that also have local state. */
+
+/* There used to be DATA statements in ANALYS.  I got rid of most of */
+/* them, and added a local logical variable FIRST that calls the entry */
+/* INITANALYS on the first call to ANALYS.  This is just so that one need */
+/* not remember to call INITANALYS first in order for the state to be */
+/* initialized. */
+
+/* Revision 1.2  1996/03/11  23:29:32  jaf */
+/* Added several comments with my own personal questions about the */
+/* Fortran 77 meaning of the parameters passed to the subroutine PREEMP. */
+
+/* Revision 1.1  1996/02/07  14:42:29  jaf */
+/* Initial revision */
+
+
+/* ****************************************************************** */
+
+/* SUBROUTINE ANALYS */
+
+/* Input: */
+/*  SPEECH */
+/*       Indices 1 through LFRAME read. */
+/* Output: */
+/*  VOICE */
+/*       Indices 1 through 2 written. */
+/*  PITCH */
+/*       Written in subroutine DYPTRK, and then perhaps read and written */
+/*       some more. */
+/*  RMS */
+/*       Written. */
+/*  RC */
+/*       Indices 1 through ORDER written (ORDER defined in contrl.fh). */
+
+/* This subroutine maintains local state from one call to the next.  If */
+/* you want to switch to using a new audio stream for this filter, or */
+/* reinitialize its state for any other reason, call the ENTRY */
+/* INITANALYS. */
+
+
+/* ENTRY PITDEC */
+
+/* Input: */
+/*  PITCH   - Encoded pitch index */
+/* Output: */
+/*  PTAU    - Decoded pitch period */
+
+/* This entry has no local state.  It accesses a "constant" array */
+/* declared in ANALYS. */
+
+/* Subroutine */ int analys_(real *speech, integer *voice, integer 
+	*pitch, real *rms, real *rc, struct lpc10_encoder_state *st)
+{
+    /* Initialized data */
+
+    static integer tau[60] = { 20,21,22,23,24,25,26,27,28,29,30,31,32,33,34,
+	    35,36,37,38,39,40,42,44,46,48,50,52,54,56,58,60,62,64,66,68,70,72,
+	    74,76,78,80,84,88,92,96,100,104,108,112,116,120,124,128,132,136,
+	    140,144,148,152,156 };
+    static integer buflim[4] = { 181,720,25,720 };
+    static real precoef = .9375f;
+
+    /* System generated locals */
+    integer i__1;
+
+    /* Local variables */
+    real amdf[60];
+    integer half;
+    real abuf[156];
+    real *bias;
+    extern /* Subroutine */ int tbdm_(real *, integer *, integer *, integer *,
+	     real *, integer *, integer *, integer *);
+    integer *awin;
+    integer midx, ewin[6]	/* was [2][3] */;
+    real ivrc[2], temp;
+    real *zpre;
+    integer *vwin;
+    integer i__, j, lanal;
+    extern /* Subroutine */ int rcchk_(integer *, real *, real *), mload_(
+	    integer *, integer *, integer *, real *, real *, real *);
+    real *inbuf, *pebuf;
+    real *lpbuf, *ivbuf;
+    real *rcbuf;
+    integer *osbuf;
+    extern /* Subroutine */ int onset_(real *, integer *, integer *, integer *
+	    , integer *, integer *, integer *, struct lpc10_encoder_state *);
+    integer *osptr;
+    extern /* Subroutine */ placea_(integer *, integer *
+	    , integer *, integer *, integer *, integer *, integer *, integer *
+	    , integer *), dcbias_(integer *, real *, real *), placev_(integer 
+	    *, integer *, integer *, integer *, integer *, integer *, integer 
+	    *, integer *, integer *, integer *, integer *);
+    integer ipitch;
+    integer *obound;
+    extern /* Subroutine */ int preemp_(real *, real *, integer *, real *, 
+	    real *), voicin_(integer *, real *, real *, integer *, integer *, 
+	    real *, real *, integer *, real *, integer *, integer *, integer *,
+	    struct lpc10_encoder_state *);
+    integer *voibuf;
+    integer mintau;
+    real *rmsbuf;
+    extern /* Subroutine */ int lpfilt_(real *, real *, integer *, integer *),
+	     ivfilt_(real *, real *, integer *, integer *, real *), energy_(
+	    integer *, real *, real *), invert_(integer *, real *, real *, 
+	    real *);
+    integer minptr, maxptr;
+    extern /* Subroutine */ int dyptrk_(real *, integer *, integer *, integer 
+	    *, integer *, integer *, struct lpc10_encoder_state *);
+    real phi[100]	/* was [10][10] */, psi[10];
+
+/* $Log$
+ * Revision 1.4  2000/01/05 08:20:39  markster
+ * Version 0.1.5 from FTP
+ *
+/* Revision 1.2  2000/01/05 08:20:39  markster
+/* Some OSS fixes and a few lpc changes to make it actually work
+/*
+ * Revision 1.2  1996/08/20  20:16:01  jaf
+ * Removed all static local variables that were SAVE'd in the Fortran
+ * code, and put them in struct lpc10_encoder_state that is passed as an
+ * argument.
+ *
+ * Removed init function, since all initialization is now done in
+ * init_lpc10_encoder_state().
+ *
+ * Revision 1.1  1996/08/19  22:29:08  jaf
+ * Initial revision
+ * */
+/* Revision 1.3  1996/03/29  22:03:47  jaf */
+/* Removed definitions for any constants that were no longer used. */
+
+/* Revision 1.2  1996/03/26  19:34:33  jaf */
+/* Added comments indicating which constants are not needed in an */
+/* application that uses the LPC-10 coder. */
+
+/* Revision 1.1  1996/02/07  14:43:51  jaf */
+/* Initial revision */
+
+/*   LPC Configuration parameters: */
+/* Frame size, Prediction order, Pitch period */
+/*       Arguments to ANALYS */
+/* $Log$
+ * Revision 1.4  2000/01/05 08:20:39  markster
+ * Version 0.1.5 from FTP
+ *
+/* Revision 1.2  2000/01/05 08:20:39  markster
+/* Some OSS fixes and a few lpc changes to make it actually work
+/*
+ * Revision 1.2  1996/08/20  20:16:01  jaf
+ * Removed all static local variables that were SAVE'd in the Fortran
+ * code, and put them in struct lpc10_encoder_state that is passed as an
+ * argument.
+ *
+ * Removed init function, since all initialization is now done in
+ * init_lpc10_encoder_state().
+ *
+ * Revision 1.1  1996/08/19  22:29:08  jaf
+ * Initial revision
+ * */
+/* Revision 1.3  1996/03/29  22:05:55  jaf */
+/* Commented out the common block variables that are not needed by the */
+/* embedded version. */
+
+/* Revision 1.2  1996/03/26  19:34:50  jaf */
+/* Added comments indicating which constants are not needed in an */
+/* application that uses the LPC-10 coder. */
+
+/* Revision 1.1  1996/02/07  14:44:09  jaf */
+/* Initial revision */
+
+/*   LPC Processing control variables: */
+
+/* *** Read-only: initialized in setup */
+
+/*  Files for Speech, Parameter, and Bitstream Input & Output, */
+/*    and message and debug outputs. */
+
+/* Here are the only files which use these variables: */
+
+/* lpcsim.f setup.f trans.f error.f vqsetup.f */
+
+/* Many files which use fdebug are not listed, since it is only used in */
+/* those other files conditionally, to print trace statements. */
+/* 	integer fsi, fso, fpi, fpo, fbi, fbo, pbin, fmsg, fdebug */
+/*  LPC order, Frame size, Quantization rate, Bits per frame, */
+/*    Error correction */
+/* Subroutine SETUP is the only place where order is assigned a value, */
+/* and that value is 10.  It could increase efficiency 1% or so to */
+/* declare order as a constant (i.e., a Fortran PARAMETER) instead of as 
+*/
+/* a variable in a COMMON block, since it is used in many places in the */
+/* core of the coding and decoding routines.  Actually, I take that back. 
+*/
+/* At least when compiling with f2c, the upper bound of DO loops is */
+/* stored in a local variable before the DO loop begins, and then that is 
+*/
+/* compared against on each iteration. */
+/* Similarly for lframe, which is given a value of MAXFRM in SETUP. */
+/* Similarly for quant, which is given a value of 2400 in SETUP.  quant */
+/* is used in only a few places, and never in the core coding and */
+/* decoding routines, so it could be eliminated entirely. */
+/* nbits is similar to quant, and is given a value of 54 in SETUP. */
+/* corrp is given a value of .TRUE. in SETUP, and is only used in the */
+/* subroutines ENCODE and DECODE.  It doesn't affect the speed of the */
+/* coder significantly whether it is .TRUE. or .FALSE., or whether it is 
+*/
+/* a constant or a variable, since it is only examined once per frame. */
+/* Leaving it as a variable that is set to .TRUE.  seems like a good */
+/* idea, since it does enable some error-correction capability for */
+/* unvoiced frames, with no change in the coding rate, and no noticeable 
+*/
+/* quality difference in the decoded speech. */
+/* 	integer quant, nbits */
+/* *** Read/write: variables for debugging, not needed for LPC algorithm 
+*/
+
+/*  Current frame, Unstable frames, Output clip count, Max onset buffer, 
+*/
+/*    Debug listing detail level, Line count on listing page */
+
+/* nframe is not needed for an embedded LPC10 at all. */
+/* nunsfm is initialized to 0 in SETUP, and incremented in subroutine */
+/* ERROR, which is only called from RCCHK.  When LPC10 is embedded into */
+/* an application, I would recommend removing the call to ERROR in RCCHK, 
+*/
+/* and remove ERROR and nunsfm completely. */
+/* iclip is initialized to 0 in SETUP, and incremented in entry SWRITE in 
+*/
+/* sread.f.  When LPC10 is embedded into an application, one might want */
+/* to cause it to be incremented in a routine that takes the output of */
+/* SYNTHS and sends it to an audio device.  It could be optionally */
+/* displayed, for those that might want to know what it is. */
+/* maxosp is never initialized to 0 in SETUP, although it probably should 
+*/
+/* be, and it is updated in subroutine ANALYS.  I doubt that its value */
+/* would be of much interest to an application in which LPC10 is */
+/* embedded. */
+/* listl and lincnt are not needed for an embedded LPC10 at all. */
+/* 	integer nframe, nunsfm, iclip, maxosp, listl, lincnt */
+/* 	common /contrl/ fsi, fso, fpi, fpo, fbi, fbo, pbin, fmsg, fdebug */
+/* 	common /contrl/ quant, nbits */
+/* 	common /contrl/ nframe, nunsfm, iclip, maxosp, listl, lincnt */
+/*       Arguments to entry PITDEC (below) */
+/* 	Parameters/constants */
+/*  Constants */
+/*    NF =     Number of frames */
+/*    AF =     Frame in which analysis is done */
+/*    OSLEN =  Length of the onset buffer */
+/*    LTAU =   Number of pitch lags */
+/*    SBUFL, SBUFH =   Start and end index of speech buffers */
+/*    LBUFL, LBUFH =   Start and end index of LPF speech buffer */
+/*   MINWIN, MAXWIN = Min and Max length of voicing (and analysis) windows
+*/
+/*    PWLEN, PWINH, PWINL = Length, upper and lower limits of pitch window
+ */
+/*    DVWINL, DVWINH = Default lower and upper limits of voicing window */
+/*       The tables TAU and BUFLIM, and the variable PRECOEF, are not */
+/*       Fortran PARAMETER's, but they are initialized with DATA */
+/*       statements, and never modified.  Thus, they need not have SAVE */
+/*       statements for them to keep their values from one invocation to 
+*/
+/*       the next. */
+/*       Local variables that need not be saved */
+/*       Local state */
+/*  Data Buffers */
+/*    INBUF	Raw speech (with DC bias removed each frame) */
+/*    PEBUF	Preemphasized speech */
+/*    LPBUF	Low pass speech buffer */
+/*    IVBUF	Inverse filtered speech */
+/*    OSBUF	Indexes of onsets in speech buffers */
+/*    VWIN	Voicing window indices */
+/*    AWIN	Analysis window indices */
+/*    EWIN	Energy window indices */
+/*    VOIBUF	Voicing decisions on windows in VWIN */
+/*    RMSBUF	RMS energy */
+/*    RCBUF	Reflection Coefficients */
+
+/*  Pitch is handled separately from the above parameters. */
+/*  The following variables deal with pitch: */
+/*    MIDX	Encoded initial pitch estimate for analysis frame */
+/*    IPITCH	Initial pitch computed for frame AF (decoded from MIDX) */
+/*    PITCH 	The encoded pitch value (index into TAU) for the present */
+/* 		frame (delayed and smoothed by Dyptrack) */
+    /* Parameter adjustments */
+    if (speech) {
+	--speech;
+	}
+    if (voice) {
+	--voice;
+	}
+    if (rc) {
+	--rc;
+	}
+
+    /* Function Body */
+
+/*   Calculations are done on future frame due to requirements */
+/*   of the pitch tracker.  Delay RMS and RC's 2 frames to give */
+/*   current frame parameters on return. */
+/*   Update all buffers */
+
+    inbuf = &(st->inbuf[0]);
+    pebuf = &(st->pebuf[0]);
+    lpbuf = &(st->lpbuf[0]);
+    ivbuf = &(st->ivbuf[0]);
+    bias = &(st->bias);
+    osbuf = &(st->osbuf[0]);
+    osptr = &(st->osptr);
+    obound = &(st->obound[0]);
+    vwin = &(st->vwin[0]);
+    awin = &(st->awin[0]);
+    voibuf = &(st->voibuf[0]);
+    rmsbuf = &(st->rmsbuf[0]);
+    rcbuf = &(st->rcbuf[0]);
+    zpre = &(st->zpre);
+
+    i__1 = 720 - contrl_1.lframe;
+    for (i__ = 181; i__ <= i__1; ++i__) {
+	inbuf[i__ - 181] = inbuf[contrl_1.lframe + i__ - 181];
+	pebuf[i__ - 181] = pebuf[contrl_1.lframe + i__ - 181];
+    }
+    i__1 = 540 - contrl_1.lframe;
+    for (i__ = 229; i__ <= i__1; ++i__) {
+	ivbuf[i__ - 229] = ivbuf[contrl_1.lframe + i__ - 229];
+    }
+    i__1 = 720 - contrl_1.lframe;
+    for (i__ = 25; i__ <= i__1; ++i__) {
+	lpbuf[i__ - 25] = lpbuf[contrl_1.lframe + i__ - 25];
+    }
+    j = 1;
+    i__1 = (*osptr) - 1;
+    for (i__ = 1; i__ <= i__1; ++i__) {
+	if (osbuf[i__ - 1] > contrl_1.lframe) {
+	    osbuf[j - 1] = osbuf[i__ - 1] - contrl_1.lframe;
+	    ++j;
+	}
+    }
+    *osptr = j;
+    voibuf[0] = voibuf[2];
+    voibuf[1] = voibuf[3];
+    for (i__ = 1; i__ <= 2; ++i__) {
+	vwin[(i__ << 1) - 2] = vwin[(i__ + 1 << 1) - 2] - contrl_1.lframe;
+	vwin[(i__ << 1) - 1] = vwin[(i__ + 1 << 1) - 1] - contrl_1.lframe;
+	awin[(i__ << 1) - 2] = awin[(i__ + 1 << 1) - 2] - contrl_1.lframe;
+	awin[(i__ << 1) - 1] = awin[(i__ + 1 << 1) - 1] - contrl_1.lframe;
+/*       EWIN(*,J) is unused for J .NE. AF, so the following shift is 
+*/
+/*       unnecessary.  It also causes error messages when the C versio
+n */
+/*       of the code created from this by f2c is run with Purify.  It 
+*/
+/*       correctly complains that uninitialized memory is being read. 
+*/
+/* 	   EWIN(1,I) = EWIN(1,I+1) - LFRAME */
+/* 	   EWIN(2,I) = EWIN(2,I+1) - LFRAME */
+	obound[i__ - 1] = obound[i__];
+	voibuf[i__ * 2] = voibuf[(i__ + 1) * 2];
+	voibuf[(i__ << 1) + 1] = voibuf[(i__ + 1 << 1) + 1];
+	rmsbuf[i__ - 1] = rmsbuf[i__];
+	i__1 = contrl_1.order;
+	for (j = 1; j <= i__1; ++j) {
+	    rcbuf[j + i__ * 10 - 11] = rcbuf[j + (i__ + 1) * 10 - 11];
+	}
+    }
+/*   Copy input speech, scale to sign+12 bit integers */
+/*   Remove long term DC bias. */
+/*       If the average value in the frame was over 1/4096 (after current 
+*/
+/*       BIAS correction), then subtract that much more from samples in */
+/*       next frame.  If the average value in the frame was under */
+/*       -1/4096, add 1/4096 more to samples in next frame.  In all other 
+*/
+/*       cases, keep BIAS the same. */
+    temp = 0.f;
+    i__1 = contrl_1.lframe;
+    for (i__ = 1; i__ <= i__1; ++i__) {
+	inbuf[720 - contrl_1.lframe + i__ - 181] = speech[i__] * 4096.f - 
+		(*bias);
+	temp += inbuf[720 - contrl_1.lframe + i__ - 181];
+    }
+    if (temp > (real) contrl_1.lframe) {
+	*bias += 1;
+    }
+    if (temp < (real) (-contrl_1.lframe)) {
+	*bias += -1;
+    }
+/*   Place Voicing Window */
+    i__ = 721 - contrl_1.lframe;
+    preemp_(&inbuf[i__ - 181], &pebuf[i__ - 181], &contrl_1.lframe, &precoef, 
+	    zpre);
+    onset_(pebuf, osbuf, osptr, &c__10, &c__181, &c__720, &contrl_1.lframe, st);
+
+/*       MAXOSP is just a debugging variable. */
+
+/* 	MAXOSP = MAX( MAXOSP, OSPTR ) */
+
+    placev_(osbuf, osptr, &c__10, &obound[2], vwin, &c__3, &contrl_1.lframe, 
+	    &c__90, &c__156, &c__307, &c__462);
+/*        The Pitch Extraction algorithm estimates the pitch for a frame 
+*/
+/*   of speech by locating the minimum of the average magnitude difference
+ */
+/*   function (AMDF).  The AMDF operates on low-pass, inverse filtered */
+/*   speech.  (The low-pass filter is an 800 Hz, 19 tap, equiripple, FIR 
+*/
+/*   filter and the inverse filter is a 2nd-order LPC filter.)  The pitch 
+*/
+/*   estimate is later refined by dynamic programming (DYPTRK).  However, 
+*/
+/*   since some of DYPTRK's parameters are a function of the voicing */
+/*  decisions, a voicing decision must precede the final pitch estimation.
+*/
+/*   See subroutines LPFILT, IVFILT, and TBDM. */
+/*       LPFILT reads indices LBUFH-LFRAME-29 = 511 through LBUFH = 720 */
+/*       of INBUF, and writes indices LBUFH+1-LFRAME = 541 through LBUFH 
+*/
+/*       = 720 of LPBUF. */
+    lpfilt_(&inbuf[228], &lpbuf[384], &c__312, &contrl_1.lframe);
+/*       IVFILT reads indices (PWINH-LFRAME-7) = 353 through PWINH = 540 
+*/
+/*       of LPBUF, and writes indices (PWINH-LFRAME+1) = 361 through */
+/*       PWINH = 540 of IVBUF. */
+    ivfilt_(&lpbuf[204], ivbuf, &c__312, &contrl_1.lframe, ivrc);
+/*       TBDM reads indices PWINL = 229 through */
+/*       (PWINL-1)+MAXWIN+(TAU(LTAU)-TAU(1))/2 = 452 of IVBUF, and writes 
+*/
+/*       indices 1 through LTAU = 60 of AMDF. */
+    tbdm_(ivbuf, &c__156, tau, &c__60, amdf, &minptr, &maxptr, &mintau);
+/*        Voicing decisions are made for each half frame of input speech. 
+*/
+/*   An initial voicing classification is made for each half of the */
+/*   analysis frame, and the voicing decisions for the present frame */
+/*   are finalized.  See subroutine VOICIN. */
+/*        The voicing detector (VOICIN) classifies the input signal as */
+/*   unvoiced (including silence) or voiced using the AMDF windowed */
+/*   maximum-to-minimum ratio, the zero crossing rate, energy measures, */
+/*   reflection coefficients, and prediction gains. */
+/*        The pitch and voicing rules apply smoothing and isolated */
+/*   corrections to the pitch and voicing estimates and, in the process, 
+*/
+/*   introduce two frames of delay into the corrected pitch estimates and 
+*/
+/*   voicing decisions. */
+    for (half = 1; half <= 2; ++half) {
+	voicin_(&vwin[4], inbuf, lpbuf, buflim, &half, &amdf[minptr - 1], &
+		amdf[maxptr - 1], &mintau, ivrc, obound, voibuf, &c__3, st);
+    }
+/*   Find the minimum cost pitch decision over several frames */
+/*   given the current voicing decision and the AMDF array */
+    dyptrk_(amdf, &c__60, &minptr, &voibuf[7], pitch, &midx, st);
+    ipitch = tau[midx - 1];
+/*   Place spectrum analysis and energy windows */
+    placea_(&ipitch, voibuf, &obound[2], &c__3, vwin, awin, ewin, &
+	    contrl_1.lframe, &c__156);
+/*  Remove short term DC bias over the analysis window, Put result in ABUF
+*/
+    lanal = awin[5] + 1 - awin[4];
+    dcbias_(&lanal, &pebuf[awin[4] - 181], abuf);
+/*       ABUF(1:LANAL) is now defined.  It is equal to */
+/*       PEBUF(AWIN(1,AF):AWIN(2,AF)) corrected for short term DC bias. */
+/*   Compute RMS over integer number of pitch periods within the */
+/*   analysis window. */
+/*   Note that in a hardware implementation this computation may be */
+/*   simplified by using diagonal elements of PHI computed by MLOAD. */
+    i__1 = ewin[5] - ewin[4] + 1;
+    energy_(&i__1, &abuf[ewin[4] - awin[4]], &rmsbuf[2]);
+/*   Matrix load and invert, check RC's for stability */
+    mload_(&contrl_1.order, &c__1, &lanal, abuf, phi, psi);
+    invert_(&contrl_1.order, phi, psi, &rcbuf[20]);
+    rcchk_(&contrl_1.order, &rcbuf[10], &rcbuf[20]);
+/*   Set return parameters */
+    voice[1] = voibuf[2];
+    voice[2] = voibuf[3];
+    *rms = rmsbuf[0];
+    i__1 = contrl_1.order;
+    for (i__ = 1; i__ <= i__1; ++i__) {
+	rc[i__] = rcbuf[i__ - 1];
+    }
+    return 0;
+} /* analys_ */

codecs/lpc10/bsynz.c

@@ -0,0 +1,423 @@
+/*
+
+$Log$
+Revision 1.4  2000/01/05 08:20:39  markster
+Version 0.1.5 from FTP
+
+Revision 1.2  2000/01/05 08:20:39  markster
+Some OSS fixes and a few lpc changes to make it actually work
+
+ * Revision 1.2  1996/08/20  20:18:55  jaf
+ * Removed all static local variables that were SAVE'd in the Fortran
+ * code, and put them in struct lpc10_decoder_state that is passed as an
+ * argument.
+ *
+ * Removed init function, since all initialization is now done in
+ * init_lpc10_decoder_state().
+ *
+ * Revision 1.1  1996/08/19  22:32:58  jaf
+ * Initial revision
+ *
+
+*/
+
+#ifdef P_R_O_T_O_T_Y_P_E_S
+extern int bsynz_(real *coef, integer *ip, integer *iv, real *sout, real *rms, real *ratio, real *g2pass, struct lpc10_decoder_state *st);
+/* comlen contrl_ 12 */
+/*:ref: random_ 4 0 */
+#endif
+
+/*  -- translated by f2c (version 19951025).
+   You must link the resulting object file with the libraries:
+	-lf2c -lm   (in that order)
+*/
+
+#include "f2c.h"
+
+/* Common Block Declarations */
+
+extern struct {
+    integer order, lframe;
+    logical corrp;
+} contrl_;
+
+#define contrl_1 contrl_
+
+/* ***************************************************************** */
+
+/* 	BSYNZ Version 54 */
+
+/* $Log$
+ * Revision 1.4  2000/01/05 08:20:39  markster
+ * Version 0.1.5 from FTP
+ *
+/* Revision 1.2  2000/01/05 08:20:39  markster
+/* Some OSS fixes and a few lpc changes to make it actually work
+/*
+ * Revision 1.2  1996/08/20  20:18:55  jaf
+ * Removed all static local variables that were SAVE'd in the Fortran
+ * code, and put them in struct lpc10_decoder_state that is passed as an
+ * argument.
+ *
+ * Removed init function, since all initialization is now done in
+ * init_lpc10_decoder_state().
+ *
+ * Revision 1.1  1996/08/19  22:32:58  jaf
+ * Initial revision
+ * */
+/* Revision 1.4  1996/03/27  18:11:22  jaf */
+/* Changed the range of NOISE printed out in the debugging statements, */
+/* even though they are commented out.  I didn't discover this until I */
+/* tried comparing two different versions of the LPC-10 coder, each with */
+/* full tracing enabled. */
+
+/* Revision 1.3  1996/03/26  19:33:23  jaf */
+/* Commented out trace statements. */
+
+/* Revision 1.2  1996/03/20  17:12:54  jaf */
+/* Added comments about which indices of array arguments are read or */
+/* written. */
+
+/* Rearranged local variable declarations to indicate which need to be */
+/* saved from one invocation to the next.  Added entry INITBSYNZ to */
+/* reinitialize the local state variables, if desired. */
+
+/* Revision 1.1  1996/02/07 14:43:15  jaf */
+/* Initial revision */
+
+
+/* ***************************************************************** */
+
+/*   Synthesize One Pitch Epoch */
+
+/* Input: */
+/*  COEF  - Predictor coefficients */
+/*          Indices 1 through ORDER read. */
+/*  IP    - Pitch period (number of samples to synthesize) */
+/*  IV    - Voicing for the current epoch */
+/*  RMS   - Energy for the current epoch */
+/*  RATIO - Energy slope for plosives */
+/*  G2PASS- Sharpening factor for 2 pass synthesis */
+/* Output: */
+/*  SOUT  - Synthesized speech */
+/*          Indices 1 through IP written. */
+
+/* This subroutine maintains local state from one call to the next.  If */
+/* you want to switch to using a new audio stream for this filter, or */
+/* reinitialize its state for any other reason, call the ENTRY */
+/* INITBSYNZ. */
+
+/* Subroutine */ int bsynz_(real *coef, integer *ip, integer *iv, 
+	real *sout, real *rms, real *ratio, real *g2pass,
+			    struct lpc10_decoder_state *st)
+{
+    /* Initialized data */
+
+    integer *ipo;
+    real *rmso;
+    static integer kexc[25] = { 8,-16,26,-48,86,-162,294,-502,718,-728,184,
+	    672,-610,-672,184,728,718,502,294,162,86,48,26,16,8 };
+    real *exc;
+    real *exc2;
+    real *lpi1;
+    real *lpi2;
+    real *lpi3;
+    real *hpi1;
+    real *hpi2;
+    real *hpi3;
+
+    /* System generated locals */
+    integer i__1, i__2;
+    real r__1, r__2;
+
+    /* Builtin functions */
+    double sqrt(doublereal);
+
+    /* Local variables */
+    real gain, xssq;
+    integer i__, j, k;
+    real noise[166], pulse;
+    integer px;
+    real sscale;
+    extern integer random_(struct lpc10_decoder_state *);
+    real xy, sum, ssq;
+    real lpi0, hpi0;
+
+/* $Log$
+ * Revision 1.4  2000/01/05 08:20:39  markster
+ * Version 0.1.5 from FTP
+ *
+/* Revision 1.2  2000/01/05 08:20:39  markster
+/* Some OSS fixes and a few lpc changes to make it actually work
+/*
+ * Revision 1.2  1996/08/20  20:18:55  jaf
+ * Removed all static local variables that were SAVE'd in the Fortran
+ * code, and put them in struct lpc10_decoder_state that is passed as an
+ * argument.
+ *
+ * Removed init function, since all initialization is now done in
+ * init_lpc10_decoder_state().
+ *
+ * Revision 1.1  1996/08/19  22:32:58  jaf
+ * Initial revision
+ * */
+/* Revision 1.3  1996/03/29  22:03:47  jaf */
+/* Removed definitions for any constants that were no longer used. */
+
+/* Revision 1.2  1996/03/26  19:34:33  jaf */
+/* Added comments indicating which constants are not needed in an */
+/* application that uses the LPC-10 coder. */
+
+/* Revision 1.1  1996/02/07  14:43:51  jaf */
+/* Initial revision */
+
+/*   LPC Configuration parameters: */
+/* Frame size, Prediction order, Pitch period */
+/*       Arguments */
+/* $Log$
+ * Revision 1.4  2000/01/05 08:20:39  markster
+ * Version 0.1.5 from FTP
+ *
+/* Revision 1.2  2000/01/05 08:20:39  markster
+/* Some OSS fixes and a few lpc changes to make it actually work
+/*
+ * Revision 1.2  1996/08/20  20:18:55  jaf
+ * Removed all static local variables that were SAVE'd in the Fortran
+ * code, and put them in struct lpc10_decoder_state that is passed as an
+ * argument.
+ *
+ * Removed init function, since all initialization is now done in
+ * init_lpc10_decoder_state().
+ *
+ * Revision 1.1  1996/08/19  22:32:58  jaf
+ * Initial revision
+ * */
+/* Revision 1.3  1996/03/29  22:05:55  jaf */
+/* Commented out the common block variables that are not needed by the */
+/* embedded version. */
+
+/* Revision 1.2  1996/03/26  19:34:50  jaf */
+/* Added comments indicating which constants are not needed in an */
+/* application that uses the LPC-10 coder. */
+
+/* Revision 1.1  1996/02/07  14:44:09  jaf */
+/* Initial revision */
+
+/*   LPC Processing control variables: */
+
+/* *** Read-only: initialized in setup */
+
+/*  Files for Speech, Parameter, and Bitstream Input & Output, */
+/*    and message and debug outputs. */
+
+/* Here are the only files which use these variables: */
+
+/* lpcsim.f setup.f trans.f error.f vqsetup.f */
+
+/* Many files which use fdebug are not listed, since it is only used in */
+/* those other files conditionally, to print trace statements. */
+/* 	integer fsi, fso, fpi, fpo, fbi, fbo, pbin, fmsg, fdebug */
+/*  LPC order, Frame size, Quantization rate, Bits per frame, */
+/*    Error correction */
+/* Subroutine SETUP is the only place where order is assigned a value, */
+/* and that value is 10.  It could increase efficiency 1% or so to */
+/* declare order as a constant (i.e., a Fortran PARAMETER) instead of as 
+*/
+/* a variable in a COMMON block, since it is used in many places in the */
+/* core of the coding and decoding routines.  Actually, I take that back. 
+*/
+/* At least when compiling with f2c, the upper bound of DO loops is */
+/* stored in a local variable before the DO loop begins, and then that is 
+*/
+/* compared against on each iteration. */
+/* Similarly for lframe, which is given a value of MAXFRM in SETUP. */
+/* Similarly for quant, which is given a value of 2400 in SETUP.  quant */
+/* is used in only a few places, and never in the core coding and */
+/* decoding routines, so it could be eliminated entirely. */
+/* nbits is similar to quant, and is given a value of 54 in SETUP. */
+/* corrp is given a value of .TRUE. in SETUP, and is only used in the */
+/* subroutines ENCODE and DECODE.  It doesn't affect the speed of the */
+/* coder significantly whether it is .TRUE. or .FALSE., or whether it is 
+*/
+/* a constant or a variable, since it is only examined once per frame. */
+/* Leaving it as a variable that is set to .TRUE.  seems like a good */
+/* idea, since it does enable some error-correction capability for */
+/* unvoiced frames, with no change in the coding rate, and no noticeable 
+*/
+/* quality difference in the decoded speech. */
+/* 	integer quant, nbits */
+/* *** Read/write: variables for debugging, not needed for LPC algorithm 
+*/
+
+/*  Current frame, Unstable frames, Output clip count, Max onset buffer, 
+*/
+/*    Debug listing detail level, Line count on listing page */
+
+/* nframe is not needed for an embedded LPC10 at all. */
+/* nunsfm is initialized to 0 in SETUP, and incremented in subroutine */
+/* ERROR, which is only called from RCCHK.  When LPC10 is embedded into */
+/* an application, I would recommend removing the call to ERROR in RCCHK, 
+*/
+/* and remove ERROR and nunsfm completely. */
+/* iclip is initialized to 0 in SETUP, and incremented in entry SWRITE in 
+*/
+/* sread.f.  When LPC10 is embedded into an application, one might want */
+/* to cause it to be incremented in a routine that takes the output of */
+/* SYNTHS and sends it to an audio device.  It could be optionally */
+/* displayed, for those that might want to know what it is. */
+/* maxosp is never initialized to 0 in SETUP, although it probably should 
+*/
+/* be, and it is updated in subroutine ANALYS.  I doubt that its value */
+/* would be of much interest to an application in which LPC10 is */
+/* embedded. */
+/* listl and lincnt are not needed for an embedded LPC10 at all. */
+/* 	integer nframe, nunsfm, iclip, maxosp, listl, lincnt */
+/* 	common /contrl/ fsi, fso, fpi, fpo, fbi, fbo, pbin, fmsg, fdebug */
+/* 	common /contrl/ quant, nbits */
+/* 	common /contrl/ nframe, nunsfm, iclip, maxosp, listl, lincnt */
+/*       Function return value definitions */
+/* 	Parameters/constants */
+/*       KEXC is not a Fortran PARAMETER, but it is an array initialized 
+*/
+/*       with a DATA statement that is never modified. */
+/*       Local variables that need not be saved */
+/*       NOISE is declared with range (1:MAXPIT+MAXORD), but only indices 
+*/
+/*       ORDER+1 through ORDER+IP are ever used, and I think that IP */
+/*       .LE. MAXPIT.  Why not declare it to be in the range (1:MAXPIT) */
+/*       and use that range? */
+/*       Local state */
+/*       I believe that only indices 1 through ORDER of EXC need to be */
+/*       saved from one invocation to the next, but we may as well save */
+/*       the whole array. */
+/*       None of these local variables were given initial values in the */
+/*       original code.  I'm guessing that 0 is a reasonable initial */
+/*       value for all of them. */
+    /* Parameter adjustments */
+    if (coef) {
+	--coef;
+	}
+    if (sout) {
+	--sout;
+	}
+
+    /* Function Body */
+    ipo = &(st->ipo);
+    exc = &(st->exc[0]);
+    exc2 = &(st->exc2[0]);
+    lpi1 = &(st->lpi1);
+    lpi2 = &(st->lpi2);
+    lpi3 = &(st->lpi3);
+    hpi1 = &(st->hpi1);
+    hpi2 = &(st->hpi2);
+    hpi3 = &(st->hpi3);
+    rmso = &(st->rmso_bsynz);
+
+/*                  MAXPIT+MAXORD=166 */
+/*  Calculate history scale factor XY and scale filter state */
+/* Computing MIN */
+    r__1 = *rmso / (*rms + 1e-6f);
+    xy = min(r__1,8.f);
+    *rmso = *rms;
+    i__1 = contrl_1.order;
+    for (i__ = 1; i__ <= i__1; ++i__) {
+	exc2[i__ - 1] = exc2[*ipo + i__ - 1] * xy;
+    }
+    *ipo = *ip;
+    if (*iv == 0) {
+/*  Generate white noise for unvoiced */
+	i__1 = *ip;
+	for (i__ = 1; i__ <= i__1; ++i__) {
+	    exc[contrl_1.order + i__ - 1] = (real) (random_(st) / 64);
+	}
+/*  Impulse doublet excitation for plosives */
+/*       (RANDOM()+32768) is in the range 0 to 2**16-1.  Therefore the
+ */
+/*       following expression should be evaluated using integers with 
+at */
+/*       least 32 bits (16 isn't enough), and PX should be in the rang
+e */
+/*       ORDER+1+0 through ORDER+1+(IP-2) .EQ. ORDER+IP-1. */
+	px = (random_(st) + 32768) * (*ip - 1) / 65536 + contrl_1.order + 1;
+	r__1 = *ratio / 4 * 1.f;
+	pulse = r__1 * 342;
+	if (pulse > 2e3f) {
+	    pulse = 2e3f;
+	}
+	exc[px - 1] += pulse;
+	exc[px] -= pulse;
+/*  Load voiced excitation */
+    } else {
+	sscale = sqrt((real) (*ip)) / 6.928f;
+	i__1 = *ip;
+	for (i__ = 1; i__ <= i__1; ++i__) {
+	    exc[contrl_1.order + i__ - 1] = 0.f;
+	    if (i__ <= 25) {
+		exc[contrl_1.order + i__ - 1] = sscale * kexc[i__ - 1];
+	    }
+	    lpi0 = exc[contrl_1.order + i__ - 1];
+	    r__2 = exc[contrl_1.order + i__ - 1] * .125f + *lpi1 * .75f;
+	    r__1 = r__2 + *lpi2 * .125f;
+	    exc[contrl_1.order + i__ - 1] = r__1 + *lpi3 * 0.f;
+	    *lpi3 = *lpi2;
+	    *lpi2 = *lpi1;
+	    *lpi1 = lpi0;
+	}
+	i__1 = *ip;
+	for (i__ = 1; i__ <= i__1; ++i__) {
+	    noise[contrl_1.order + i__ - 1] = random_(st) * 1.f / 64;
+	    hpi0 = noise[contrl_1.order + i__ - 1];
+	    r__2 = noise[contrl_1.order + i__ - 1] * -.125f + *hpi1 * .25f;
+	    r__1 = r__2 + *hpi2 * -.125f;
+	    noise[contrl_1.order + i__ - 1] = r__1 + *hpi3 * 0.f;
+	    *hpi3 = *hpi2;
+	    *hpi2 = *hpi1;
+	    *hpi1 = hpi0;
+	}
+	i__1 = *ip;
+	for (i__ = 1; i__ <= i__1; ++i__) {
+	    exc[contrl_1.order + i__ - 1] += noise[contrl_1.order + i__ - 1];
+	}
+    }
+/*   Synthesis filters: */
+/*    Modify the excitation with all-zero filter  1 + G*SUM */
+    xssq = 0.f;
+    i__1 = *ip;
+    for (i__ = 1; i__ <= i__1; ++i__) {
+	k = contrl_1.order + i__;
+	sum = 0.f;
+	i__2 = contrl_1.order;
+	for (j = 1; j <= i__2; ++j) {
+	    sum += coef[j] * exc[k - j - 1];
+	}
+	sum *= *g2pass;
+	exc2[k - 1] = sum + exc[k - 1];
+    }
+/*   Synthesize using the all pole filter  1 / (1 - SUM) */
+    i__1 = *ip;
+    for (i__ = 1; i__ <= i__1; ++i__) {
+	k = contrl_1.order + i__;
+	sum = 0.f;
+	i__2 = contrl_1.order;
+	for (j = 1; j <= i__2; ++j) {
+	    sum += coef[j] * exc2[k - j - 1];
+	}
+	exc2[k - 1] = sum + exc2[k - 1];
+	xssq += exc2[k - 1] * exc2[k - 1];
+    }
+/*  Save filter history for next epoch */
+    i__1 = contrl_1.order;
+    for (i__ = 1; i__ <= i__1; ++i__) {
+	exc[i__ - 1] = exc[*ip + i__ - 1];
+	exc2[i__ - 1] = exc2[*ip + i__ - 1];
+    }
+/*  Apply gain to match RMS */
+    r__1 = *rms * *rms;
+    ssq = r__1 * *ip;
+    gain = sqrt(ssq / xssq);
+    i__1 = *ip;
+    for (i__ = 1; i__ <= i__1; ++i__) {
+	sout[i__] = gain * exc2[contrl_1.order + i__ - 1];
+    }
+    return 0;
+} /* bsynz_ */

codecs/lpc10/chanwr.c

@@ -0,0 +1,225 @@
+/*
+
+$Log$
+Revision 1.4  2000/01/05 08:20:39  markster
+Version 0.1.5 from FTP
+
+Revision 1.2  2000/01/05 08:20:39  markster
+Some OSS fixes and a few lpc changes to make it actually work
+
+ * Revision 1.2  1996/08/20  20:20:24  jaf
+ * Removed all static local variables that were SAVE'd in the Fortran
+ * code, and put them in struct lpc10_encoder_state that is passed as an
+ * argument.
+ *
+ * Revision 1.1  1996/08/19  22:40:31  jaf
+ * Initial revision
+ *
+
+*/
+
+#ifdef P_R_O_T_O_T_Y_P_E_S
+extern int chanwr_(integer *order, integer *ipitv, integer *irms, integer *irc, integer *ibits, struct lpc10_encoder_state *st);
+extern int chanrd_(integer *order, integer *ipitv, integer *irms, integer *irc, integer *ibits);
+#endif
+
+/*  -- translated by f2c (version 19951025).
+   You must link the resulting object file with the libraries:
+	-lf2c -lm   (in that order)
+*/
+
+#include "f2c.h"
+
+/* *********************************************************************** */
+
+/* 	CHANL Version 49 */
+
+/* $Log$
+ * Revision 1.4  2000/01/05 08:20:39  markster
+ * Version 0.1.5 from FTP
+ *
+/* Revision 1.2  2000/01/05 08:20:39  markster
+/* Some OSS fixes and a few lpc changes to make it actually work
+/*
+ * Revision 1.2  1996/08/20  20:20:24  jaf
+ * Removed all static local variables that were SAVE'd in the Fortran
+ * code, and put them in struct lpc10_encoder_state that is passed as an
+ * argument.
+ *
+ * Revision 1.1  1996/08/19  22:40:31  jaf
+ * Initial revision
+ * */
+/* Revision 1.3  1996/03/21  15:14:57  jaf */
+/* Added comments about which indices of argument arrays are read or */
+/* written, and about the one bit of local state in CHANWR.  CHANRD */
+/* has no local state. */
+
+/* Revision 1.2  1996/03/13  18:55:10  jaf */
+/* Comments added explaining which of the local variables of this */
+/* subroutine need to be saved from one invocation to the next, and which */
+/* do not. */
+
+/* Revision 1.1  1996/02/07 14:43:31  jaf */
+/* Initial revision */
+
+
+/* *********************************************************************** */
+
+/* CHANWR: */
+/*   Place quantized parameters into bitstream */
+
+/* Input: */
+/*  ORDER  - Number of reflection coefficients (not really variable) */
+/*  IPITV  - Quantized pitch/voicing parameter */
+/*  IRMS   - Quantized energy parameter */
+/*  IRC    - Quantized reflection coefficients */
+/*           Indices 1 through ORDER read. */
+/* Output: */
+/*  IBITS  - Serial bitstream */
+/*           Indices 1 through 54 written. */
+/*           Bit 54, the SYNC bit, alternates from one call to the next. */
+
+/* Subroutine CHANWR maintains one bit of local state from one call to */
+/* the next, in the variable ISYNC.  I believe that this one bit is only */
+/* intended to allow a receiver to resynchronize its interpretation of */
+/* the bit stream, by looking for which of the 54 bits alternates every */
+/* frame time.  This is just a simple framing mechanism that is not */
+/* useful when other, higher overhead framing mechanisms are used to */
+/* transmit the coded frames. */
+
+/* I'm not going to make an entry to reinitialize this bit, since it */
+/* doesn't help a receiver much to know whether the first sync bit is a 0 */
+/* or a 1.  It needs to examine several frames in sequence to have */
+/* reasonably good assurance that its framing is correct. */
+
+
+/* CHANRD: */
+/*   Reconstruct parameters from bitstream */
+
+/* Input: */
+/*  ORDER  - Number of reflection coefficients (not really variable) */
+/*  IBITS  - Serial bitstream */
+/*           Indices 1 through 53 read (SYNC bit is ignored). */
+/* Output: */
+/*  IPITV  - Quantized pitch/voicing parameter */
+/*  IRMS   - Quantized energy parameter */
+/*  IRC    - Quantized reflection coefficients */
+/*           Indices 1 through ORDER written */
+
+/* Entry CHANRD has no local state. */
+
+
+
+/*   IBITS is 54 bits of LPC data ordered as follows: */
+/* 	R1-0, R2-0, R3-0,  P-0,  A-0, */
+/* 	R1-1, R2-1, R3-1,  P-1,  A-1, */
+/* 	R1-2, R4-0, R3-2,  A-2,  P-2, R4-1, */
+/* 	R1-3, R2-2, R3-3, R4-2,  A-3, */
+/* 	R1-4, R2-3, R3-4, R4-3,  A-4, */
+/* 	 P-3, R2-4, R7-0, R8-0,  P-4, R4-4, */
+/* 	R5-0, R6-0, R7-1,R10-0, R8-1, */
+/* 	R5-1, R6-1, R7-2, R9-0,  P-5, */
+/* 	R5-2, R6-2,R10-1, R8-2,  P-6, R9-1, */
+/* 	R5-3, R6-3, R7-3, R9-2, R8-3, SYNC */
+/* Subroutine */ int chanwr_0_(int n__, integer *order, integer *ipitv, 
+	integer *irms, integer *irc, integer *ibits,
+			       struct lpc10_encoder_state *st)
+{
+    /* Initialized data */
+
+    integer *isync;
+    static integer bit[10] = { 2,4,8,8,8,8,16,16,16,16 };
+    static integer iblist[53] = { 13,12,11,1,2,13,12,11,1,2,13,10,11,2,1,10,
+	    13,12,11,10,2,13,12,11,10,2,1,12,7,6,1,10,9,8,7,4,6,9,8,7,5,1,9,8,
+	    4,6,1,5,9,8,7,5,6 };
+
+    /* System generated locals */
+    integer i__1;
+
+    /* Local variables */
+    integer itab[13], i__;
+
+/*       Arguments */
+/*       Parameters/constants */
+/*       These arrays are not Fortran PARAMETER's, but they are defined */
+/*       by DATA statements below, and their contents are never altered. 
+*/
+/*       Local variables that need not be saved */
+/*       Local state */
+/*       ISYNC is only used by CHANWR, not by ENTRY CHANRD. */
+
+    /* Parameter adjustments */
+    --irc;
+    --ibits;
+
+    /* Function Body */
+    switch(n__) {
+	case 1: goto L_chanrd;
+	}
+
+    isync = &(st->isync);
+
+/* ***********************************************************************
+ */
+/* 	Place quantized parameters into bitstream */
+/* ***********************************************************************
+ */
+/*   Place parameters into ITAB */
+    itab[0] = *ipitv;
+    itab[1] = *irms;
+    itab[2] = 0;
+    i__1 = *order;
+    for (i__ = 1; i__ <= i__1; ++i__) {
+	itab[i__ + 2] = irc[*order + 1 - i__] & 32767;
+    }
+/*   Put 54 bits into IBITS array */
+    for (i__ = 1; i__ <= 53; ++i__) {
+	ibits[i__] = itab[iblist[i__ - 1] - 1] & 1;
+	itab[iblist[i__ - 1] - 1] /= 2;
+    }
+    ibits[54] = *isync & 1;
+    *isync = 1 - *isync;
+    return 0;
+/* ***********************************************************************
+ */
+/* 	Reconstruct parameters from bitstream */
+/* ***********************************************************************
+ */
+
+L_chanrd:
+/*   Reconstruct ITAB */
+    for (i__ = 1; i__ <= 13; ++i__) {
+	itab[i__ - 1] = 0;
+    }
+    for (i__ = 1; i__ <= 53; ++i__) {
+	itab[iblist[54 - i__ - 1] - 1] = (itab[iblist[54 - i__ - 1] - 1] << 1)
+		 + ibits[54 - i__];
+    }
+/*   Sign extend RC's */
+    i__1 = *order;
+    for (i__ = 1; i__ <= i__1; ++i__) {
+	if ((itab[i__ + 2] & bit[i__ - 1]) != 0) {
+	    itab[i__ + 2] -= bit[i__ - 1] << 1;
+	}
+    }
+/*   Restore variables */
+    *ipitv = itab[0];
+    *irms = itab[1];
+    i__1 = *order;
+    for (i__ = 1; i__ <= i__1; ++i__) {
+	irc[i__] = itab[*order + 4 - i__ - 1];
+    }
+    return 0;
+} /* chanwr_ */
+
+/* Subroutine */ int chanwr_(integer *order, integer *ipitv, integer *irms, 
+	integer *irc, integer *ibits, struct lpc10_encoder_state *st)
+{
+    return chanwr_0_(0, order, ipitv, irms, irc, ibits, st);
+    }
+
+/* Subroutine */ int chanrd_(integer *order, integer *ipitv, integer *irms, 
+	integer *irc, integer *ibits)
+{
+    return chanwr_0_(1, order, ipitv, irms, irc, ibits, 0);
+    }

codecs/lpc10/dcbias.c

@@ -0,0 +1,95 @@
+/*
+
+$Log$
+Revision 1.4  2000/01/05 08:20:39  markster
+Version 0.1.5 from FTP
+
+Revision 1.2  2000/01/05 08:20:39  markster
+Some OSS fixes and a few lpc changes to make it actually work
+
+ * Revision 1.1  1996/08/19  22:40:23  jaf
+ * Initial revision
+ *
+
+*/
+
+#ifdef P_R_O_T_O_T_Y_P_E_S
+extern int dcbias_(integer *len, real *speech, real *sigout);
+#endif
+
+/*  -- translated by f2c (version 19951025).
+   You must link the resulting object file with the libraries:
+	-lf2c -lm   (in that order)
+*/
+
+#include "f2c.h"
+
+/* ********************************************************************* */
+
+/* 	DCBIAS Version 50 */
+
+/* $Log$
+ * Revision 1.4  2000/01/05 08:20:39  markster
+ * Version 0.1.5 from FTP
+ *
+/* Revision 1.2  2000/01/05 08:20:39  markster
+/* Some OSS fixes and a few lpc changes to make it actually work
+/*
+ * Revision 1.1  1996/08/19  22:40:23  jaf
+ * Initial revision
+ * */
+/* Revision 1.3  1996/03/18  21:19:22  jaf */
+/* Just added a few comments about which array indices of the arguments */
+/* are used, and mentioning that this subroutine has no local state. */
+
+/* Revision 1.2  1996/03/13  16:44:53  jaf */
+/* Comments added explaining that none of the local variables of this */
+/* subroutine need to be saved from one invocation to the next. */
+
+/* Revision 1.1  1996/02/07 14:44:21  jaf */
+/* Initial revision */
+
+
+/* ********************************************************************* */
+
+/* Calculate and remove DC bias from buffer. */
+
+/* Input: */
+/*  LEN    - Length of speech buffers */
+/*  SPEECH - Input speech buffer */
+/*           Indices 1 through LEN read. */
+/* Output: */
+/*  SIGOUT - Output speech buffer */
+/*           Indices 1 through LEN written */
+
+/* This subroutine has no local state. */
+
+/* Subroutine */ int dcbias_(integer *len, real *speech, real *sigout)
+{
+    /* System generated locals */
+    integer i__1;
+
+    /* Local variables */
+    real bias;
+    integer i__;
+
+/* 	Arguments */
+/*       Local variables that need not be saved */
+    /* Parameter adjustments */
+    --sigout;
+    --speech;
+
+    /* Function Body */
+    bias = 0.f;
+    i__1 = *len;
+    for (i__ = 1; i__ <= i__1; ++i__) {
+	bias += speech[i__];
+    }
+    bias /= *len;
+    i__1 = *len;
+    for (i__ = 1; i__ <= i__1; ++i__) {
+	sigout[i__] = speech[i__] - bias;
+    }
+    return 0;
+} /* dcbias_ */
+

codecs/lpc10/decode.c

@@ -0,0 +1,589 @@
+/*
+
+$Log$
+Revision 1.4  2000/01/05 08:20:39  markster
+Version 0.1.5 from FTP
+
+Revision 1.2  2000/01/05 08:20:39  markster
+Some OSS fixes and a few lpc changes to make it actually work
+
+ * Revision 1.2  1996/08/20  20:22:39  jaf
+ * Removed all static local variables that were SAVE'd in the Fortran
+ * code, and put them in struct lpc10_decoder_state that is passed as an
+ * argument.
+ *
+ * Removed init function, since all initialization is now done in
+ * init_lpc10_decoder_state().
+ *
+ * Revision 1.1  1996/08/19  22:32:38  jaf
+ * Initial revision
+ *
+
+*/
+
+#ifdef P_R_O_T_O_T_Y_P_E_S
+extern int decode_(integer *ipitv, integer *irms, integer *irc, integer *voice, integer *pitch, real *rms, real *rc, struct lpc10_decoder_state *st);
+/* comlen contrl_ 12 */
+/*:ref: ham84_ 14 3 4 4 4 */
+/*:ref: median_ 4 3 4 4 4 */
+#endif
+
+/*  -- translated by f2c (version 19951025).
+   You must link the resulting object file with the libraries:
+	-lf2c -lm   (in that order)
+*/
+
+#include "f2c.h"
+
+/* Common Block Declarations */
+
+extern struct {
+    integer order, lframe;
+    logical corrp;
+} contrl_;
+
+#define contrl_1 contrl_
+
+/* Table of constant values */
+
+static integer c__2 = 2;
+
+/* ***************************************************************** */
+
+/* 	DECODE Version 54 */
+
+/* $Log$
+ * Revision 1.4  2000/01/05 08:20:39  markster
+ * Version 0.1.5 from FTP
+ *
+/* Revision 1.2  2000/01/05 08:20:39  markster
+/* Some OSS fixes and a few lpc changes to make it actually work
+/*
+ * Revision 1.2  1996/08/20  20:22:39  jaf
+ * Removed all static local variables that were SAVE'd in the Fortran
+ * code, and put them in struct lpc10_decoder_state that is passed as an
+ * argument.
+ *
+ * Removed init function, since all initialization is now done in
+ * init_lpc10_decoder_state().
+ *
+ * Revision 1.1  1996/08/19  22:32:38  jaf
+ * Initial revision
+ * */
+/* Revision 1.5  1996/05/23  20:06:03  jaf */
+/* Assigned PITCH a "default" value on the first call, since otherwise it */
+/* would be left uninitialized. */
+
+/* Revision 1.4  1996/03/26  19:35:18  jaf */
+/* Commented out trace statements. */
+
+/* Revision 1.3  1996/03/21  21:10:50  jaf */
+/* Added entry INITDECODE to reinitialize the local state of subroutine */
+/* DECODE. */
+
+/* Revision 1.2  1996/03/21  21:04:50  jaf */
+/* Determined which local variables should be saved from one invocation */
+/* to the next, and guessed initial values for some that should have been */
+/* saved, but weren't given initial values.  Many of the arrays are */
+/* "constants", and many local variables are only used if the "global" */
+/* variable CORRP is .TRUE. */
+
+/* Added comments explaining which indices of array arguments are read or */
+/* written. */
+
+/* Revision 1.1  1996/02/12 03:21:10  jaf */
+/* Initial revision */
+
+
+/* ***************************************************************** */
+
+/*   This subroutine provides error correction and decoding */
+/*   for all LPC parameters */
+
+/* Input: */
+/*  IPITV  - Index value of pitch */
+/*  IRMS   - Coded Energy */
+/*  CORRP  - Error correction: */
+/*    If FALSE, parameters are decoded directly with no delay.  If TRUE, */
+/*    most important parameter bits are protected by Hamming code and */
+/*    median smoothed.  This requires an additional frame of delay. */
+/* Input/Output: */
+/*  IRC    - Coded Reflection Coefficients */
+/*           Indices 1 through ORDER always read, then written. */
+/* Output: */
+/*  VOICE  - Half frame voicing decisions */
+/*           Indices 1 through 2 written. */
+/*  PITCH  - Decoded pitch */
+/*  RMS    - Energy */
+/*  RC     - Reflection coefficients */
+/*           Indices 1 through ORDER written. */
+
+/*  NOTE: Zero RC's should be done more directly, but this would affect */
+/*   coded parameter printout. */
+
+/* This subroutine maintains local state from one call to the next.  If */
+/* you want to switch to using a new audio stream for this filter, or */
+/* reinitialize its state for any other reason, call the ENTRY */
+/* INITDECODE. */
+
+/* Subroutine */ int decode_(integer *ipitv, integer *irms, 
+	integer *irc, integer *voice, integer *pitch, real *rms, real *rc,
+			     struct lpc10_decoder_state *st)
+{
+    /* Initialized data */
+
+    logical *first;
+    static integer ethrs = 2048;
+    static integer ethrs1 = 128;
+    static integer ethrs2 = 1024;
+    static integer ethrs3 = 2048;
+    static integer ivtab[32] = { 24960,24960,24960,24960,25480,25480,25483,
+	    25480,16640,1560,1560,1560,16640,1816,1563,1560,24960,24960,24859,
+	    24856,26001,25881,25915,25913,1560,1560,7800,3640,1561,1561,3643,
+	    3641 };
+    static real corth[32]	/* was [4][8] */ = { 32767.f,10.f,5.f,0.f,
+	    32767.f,8.f,4.f,0.f,32.f,6.4f,3.2f,0.f,32.f,6.4f,3.2f,0.f,32.f,
+	    11.2f,6.4f,0.f,32.f,11.2f,6.4f,0.f,16.f,5.6f,3.2f,0.f,16.f,5.6f,
+	    3.2f,0.f };
+    static integer detau[128] = { 0,0,0,3,0,3,3,31,0,3,3,21,3,3,29,30,0,3,3,
+	    20,3,25,27,26,3,23,58,22,3,24,28,3,0,3,3,3,3,39,33,32,3,37,35,36,
+	    3,38,34,3,3,42,46,44,50,40,48,3,54,3,56,3,52,3,3,1,0,3,3,108,3,78,
+	    100,104,3,84,92,88,156,80,96,3,3,74,70,72,66,76,68,3,62,3,60,3,64,
+	    3,3,1,3,116,132,112,148,152,3,3,140,3,136,3,144,3,3,1,124,120,128,
+	    3,3,3,3,1,3,3,3,1,3,1,1,1 };
+    static integer rmst[64] = { 1024,936,856,784,718,656,600,550,502,460,420,
+	    384,352,328,294,270,246,226,206,188,172,158,144,132,120,110,102,
+	    92,84,78,70,64,60,54,50,46,42,38,34,32,30,26,24,22,20,18,17,16,15,
+	    14,13,12,11,10,9,8,7,6,5,4,3,2,1,0 };
+    static integer detab7[32] = { 4,11,18,25,32,39,46,53,60,66,72,77,82,87,92,
+	    96,101,104,108,111,114,115,117,119,121,122,123,124,125,126,127,
+	    127 };
+    static real descl[8] = { .6953f,.625f,.5781f,.5469f,.5312f,.5391f,.4688f,
+	    .3828f };
+    integer *ivp2h;
+    static integer deadd[8] = { 1152,-2816,-1536,-3584,-1280,-2432,768,-1920 }
+	    ;
+    static integer qb[8] = { 511,511,1023,1023,1023,1023,2047,4095 };
+    static integer nbit[10] = { 8,8,5,5,4,4,4,4,3,2 };
+    static integer zrc[10] = { 0,0,0,0,0,3,0,2,0,0 };
+    static integer bit[5] = { 2,4,8,16,32 };
+    integer *iovoic;
+    integer *iavgp;
+    integer *iptold;
+    integer *erate;
+    integer *drc;
+    integer *dpit;
+    integer *drms;
+
+    /* System generated locals */
+    integer i__1, i__2;
+
+    /* Builtin functions */
+    integer pow_ii(integer *, integer *);
+
+    /* Local variables */
+    extern /* Subroutine */ int ham84_(integer *, integer *, integer *);
+    integer ipit, iout, i__, icorf, index, ivoic, ixcor, i1, i2, i4;
+    extern integer median_(integer *, integer *, integer *);
+    integer ishift, errcnt, lsb;
+
+/* $Log$
+ * Revision 1.4  2000/01/05 08:20:39  markster
+ * Version 0.1.5 from FTP
+ *
+/* Revision 1.2  2000/01/05 08:20:39  markster
+/* Some OSS fixes and a few lpc changes to make it actually work
+/*
+ * Revision 1.2  1996/08/20  20:22:39  jaf
+ * Removed all static local variables that were SAVE'd in the Fortran
+ * code, and put them in struct lpc10_decoder_state that is passed as an
+ * argument.
+ *
+ * Removed init function, since all initialization is now done in
+ * init_lpc10_decoder_state().
+ *
+ * Revision 1.1  1996/08/19  22:32:38  jaf
+ * Initial revision
+ * */
+/* Revision 1.3  1996/03/29  22:03:47  jaf */
+/* Removed definitions for any constants that were no longer used. */
+
+/* Revision 1.2  1996/03/26  19:34:33  jaf */
+/* Added comments indicating which constants are not needed in an */
+/* application that uses the LPC-10 coder. */
+
+/* Revision 1.1  1996/02/07  14:43:51  jaf */
+/* Initial revision */
+
+/*   LPC Configuration parameters: */
+/* Frame size, Prediction order, Pitch period */
+/*       Arguments */
+/* $Log$
+ * Revision 1.4  2000/01/05 08:20:39  markster
+ * Version 0.1.5 from FTP
+ *
+/* Revision 1.2  2000/01/05 08:20:39  markster
+/* Some OSS fixes and a few lpc changes to make it actually work
+/*
+ * Revision 1.2  1996/08/20  20:22:39  jaf
+ * Removed all static local variables that were SAVE'd in the Fortran
+ * code, and put them in struct lpc10_decoder_state that is passed as an
+ * argument.
+ *
+ * Removed init function, since all initialization is now done in
+ * init_lpc10_decoder_state().
+ *
+ * Revision 1.1  1996/08/19  22:32:38  jaf
+ * Initial revision
+ * */
+/* Revision 1.3  1996/03/29  22:05:55  jaf */
+/* Commented out the common block variables that are not needed by the */
+/* embedded version. */
+
+/* Revision 1.2  1996/03/26  19:34:50  jaf */
+/* Added comments indicating which constants are not needed in an */
+/* application that uses the LPC-10 coder. */
+
+/* Revision 1.1  1996/02/07  14:44:09  jaf */
+/* Initial revision */
+
+/*   LPC Processing control variables: */
+
+/* *** Read-only: initialized in setup */
+
+/*  Files for Speech, Parameter, and Bitstream Input & Output, */
+/*    and message and debug outputs. */
+
+/* Here are the only files which use these variables: */
+
+/* lpcsim.f setup.f trans.f error.f vqsetup.f */
+
+/* Many files which use fdebug are not listed, since it is only used in */
+/* those other files conditionally, to print trace statements. */
+/* 	integer fsi, fso, fpi, fpo, fbi, fbo, pbin, fmsg, fdebug */
+/*  LPC order, Frame size, Quantization rate, Bits per frame, */
+/*    Error correction */
+/* Subroutine SETUP is the only place where order is assigned a value, */
+/* and that value is 10.  It could increase efficiency 1% or so to */
+/* declare order as a constant (i.e., a Fortran PARAMETER) instead of as 
+*/
+/* a variable in a COMMON block, since it is used in many places in the */
+/* core of the coding and decoding routines.  Actually, I take that back. 
+*/
+/* At least when compiling with f2c, the upper bound of DO loops is */
+/* stored in a local variable before the DO loop begins, and then that is 
+*/
+/* compared against on each iteration. */
+/* Similarly for lframe, which is given a value of MAXFRM in SETUP. */
+/* Similarly for quant, which is given a value of 2400 in SETUP.  quant */
+/* is used in only a few places, and never in the core coding and */
+/* decoding routines, so it could be eliminated entirely. */
+/* nbits is similar to quant, and is given a value of 54 in SETUP. */
+/* corrp is given a value of .TRUE. in SETUP, and is only used in the */
+/* subroutines ENCODE and DECODE.  It doesn't affect the speed of the */
+/* coder significantly whether it is .TRUE. or .FALSE., or whether it is 
+*/
+/* a constant or a variable, since it is only examined once per frame. */
+/* Leaving it as a variable that is set to .TRUE.  seems like a good */
+/* idea, since it does enable some error-correction capability for */
+/* unvoiced frames, with no change in the coding rate, and no noticeable 
+*/
+/* quality difference in the decoded speech. */
+/* 	integer quant, nbits */
+/* *** Read/write: variables for debugging, not needed for LPC algorithm 
+*/
+
+/*  Current frame, Unstable frames, Output clip count, Max onset buffer, 
+*/
+/*    Debug listing detail level, Line count on listing page */
+
+/* nframe is not needed for an embedded LPC10 at all. */
+/* nunsfm is initialized to 0 in SETUP, and incremented in subroutine */
+/* ERROR, which is only called from RCCHK.  When LPC10 is embedded into */
+/* an application, I would recommend removing the call to ERROR in RCCHK, 
+*/
+/* and remove ERROR and nunsfm completely. */
+/* iclip is initialized to 0 in SETUP, and incremented in entry SWRITE in 
+*/
+/* sread.f.  When LPC10 is embedded into an application, one might want */
+/* to cause it to be incremented in a routine that takes the output of */
+/* SYNTHS and sends it to an audio device.  It could be optionally */
+/* displayed, for those that might want to know what it is. */
+/* maxosp is never initialized to 0 in SETUP, although it probably should 
+*/
+/* be, and it is updated in subroutine ANALYS.  I doubt that its value */
+/* would be of much interest to an application in which LPC10 is */
+/* embedded. */
+/* listl and lincnt are not needed for an embedded LPC10 at all. */
+/* 	integer nframe, nunsfm, iclip, maxosp, listl, lincnt */
+/* 	common /contrl/ fsi, fso, fpi, fpo, fbi, fbo, pbin, fmsg, fdebug */
+/* 	common /contrl/ quant, nbits */
+/* 	common /contrl/ nframe, nunsfm, iclip, maxosp, listl, lincnt */
+/*       Function return value definitions */
+
+/*       Parameters/constants */
+
+/*       The variables below that are not Fortran PARAMETER's are */
+/*       initialized with DATA statements, and then never modified. */
+/*       The following are used regardless of CORRP's value. */
+
+/*       DETAU, NBIT, QB, DEADD, DETAB7, RMST, DESCL */
+
+/*       The following are used only if CORRP is .TRUE. */
+
+/*       ETHRS, ETHRS1, ETHRS2, ETHRS3, IVTAB, BIT, CORTH, ZRC */
+
+/*       Local variables that need not be saved */
+
+/*       The following are used regardless of CORRP's value */
+/*       The following are used only if CORRP is .TRUE. */
+
+/*       Local state */
+
+/*       The following are used regardless of CORRP's value */
+/*       The following are used only if CORRP is .TRUE. */
+/*       I am guessing the initial values for IVP2H, IOVOIC, DRC, DPIT, */
+/*       and DRMS.  They should be checked to see if they are reasonable. 
+*/
+/*       I'm also guessing for ERATE, but I think 0 is the right initial 
+*/
+/*       value. */
+    /* Parameter adjustments */
+    if (irc) {
+	--irc;
+	}
+    if (voice) {
+	--voice;
+	}
+    if (rc) {
+	--rc;
+	}
+
+    /* Function Body */
+
+    iptold = &(st->iptold);
+    first = &(st->first);
+    ivp2h = &(st->ivp2h);
+    iovoic = &(st->iovoic);
+    iavgp = &(st->iavgp);
+    erate = &(st->erate);
+    drc = &(st->drc[0]);
+    dpit = &(st->dpit[0]);
+    drms = &(st->drms[0]);
+
+/* DATA statements for "constants" defined above. */
+/* 	IF (LISTL.GE.3) WRITE(FDEBUG,800) IPITV,IRMS,(IRC(J),J=1,ORDER) */
+/* 800	FORMAT(1X,' <<ERRCOR IN>>',T32,6X,I6,I5,T50,10I8) */
+/*  If no error correction, do pitch and voicing then jump to decode */
+    i4 = detau[*ipitv];
+    if (! contrl_1.corrp) {
+	voice[1] = 1;
+	voice[2] = 1;
+	if (*ipitv <= 1) {
+	    voice[1] = 0;
+	}
+	if (*ipitv == 0 || *ipitv == 2) {
+	    voice[2] = 0;
+	}
+	*pitch = i4;
+	if (*pitch <= 4) {
+	    *pitch = *iptold;
+	}
+	if (voice[1] == 1 && voice[2] == 1) {
+	    *iptold = *pitch;
+	}
+	if (voice[1] != voice[2]) {
+	    *pitch = *iptold;
+	}
+	goto L900;
+    }
+/*  Do error correction pitch and voicing */
+    if (i4 > 4) {
+	dpit[0] = i4;
+	ivoic = 2;
+	*iavgp = (*iavgp * 15 + i4 + 8) / 16;
+    } else {
+	ivoic = i4;
+	dpit[0] = *iavgp;
+    }
+    drms[0] = *irms;
+    i__1 = contrl_1.order;
+    for (i__ = 1; i__ <= i__1; ++i__) {
+	drc[i__ * 3 - 3] = irc[i__];
+    }
+/*  Determine index to IVTAB from V/UV decision */
+/*  If error rate is high then use alternate table */
+    index = (*ivp2h << 4) + (*iovoic << 2) + ivoic + 1;
+    i1 = ivtab[index - 1];
+    ipit = i1 & 3;
+    icorf = i1 / 8;
+    if (*erate < ethrs) {
+	icorf /= 64;
+    }
+/*  Determine error rate:  4=high    1=low */
+    ixcor = 4;
+    if (*erate < ethrs3) {
+	ixcor = 3;
+    }
+    if (*erate < ethrs2) {
+	ixcor = 2;
+    }
+    if (*erate < ethrs1) {
+	ixcor = 1;
+    }
+/*  Voice/unvoice decision determined from bits 0 and 1 of IVTAB */
+    voice[1] = icorf / 2 & 1;
+    voice[2] = icorf & 1;
+/*  Skip decoding on first frame because present data not yet available */
+    if (*first) {
+	*first = FALSE_;
+/*          Assign PITCH a "default" value on the first call, since */
+/*          otherwise it would be left uninitialized.  The two lines 
+*/
+/*          below were copied from above, since it seemed like a */
+/*          reasonable thing to do for the first call. */
+	*pitch = i4;
+	if (*pitch <= 4) {
+	    *pitch = *iptold;
+	}
+	goto L500;
+    }
+/*  If bit 4 of ICORF is set then correct RMS and RC(1) - RC(4). */
+/*    Determine error rate and correct errors using a Hamming 8,4 code */
+/*    during transition or unvoiced frame.  If IOUT is negative, */
+/*    more than 1 error occurred, use previous frame's parameters. */
+    if ((icorf & bit[3]) != 0) {
+	errcnt = 0;
+	lsb = drms[1] & 1;
+	index = (drc[22] << 4) + drms[1] / 2;
+	ham84_(&index, &iout, &errcnt);
+	drms[1] = drms[2];
+	if (iout >= 0) {
+	    drms[1] = (iout << 1) + lsb;
+	}
+	for (i__ = 1; i__ <= 4; ++i__) {
+	    if (i__ == 1) {
+		i1 = ((drc[25] & 7) << 1) + (drc[28] & 1);
+	    } else {
+		i1 = drc[(9 - i__) * 3 - 2] & 15;
+	    }
+	    i2 = drc[(5 - i__) * 3 - 2] & 31;
+	    lsb = i2 & 1;
+	    index = (i1 << 4) + i2 / 2;
+	    ham84_(&index, &iout, &errcnt);
+	    if (iout >= 0) {
+		iout = (iout << 1) + lsb;
+		if ((iout & 16) == 16) {
+		    iout += -32;
+		}
+	    } else {
+		iout = drc[(5 - i__) * 3 - 1];
+	    }
+	    drc[(5 - i__) * 3 - 2] = iout;
+	}
+/*  Determine error rate */
+	*erate = *erate * .96875f + errcnt * 102;
+    }
+/*  Get unsmoothed RMS, RC's, and PITCH */
+    *irms = drms[1];
+    i__1 = contrl_1.order;
+    for (i__ = 1; i__ <= i__1; ++i__) {
+	irc[i__] = drc[i__ * 3 - 2];
+    }
+    if (ipit == 1) {
+	dpit[1] = dpit[2];
+    }
+    if (ipit == 3) {
+	dpit[1] = dpit[0];
+    }
+    *pitch = dpit[1];
+/*  If bit 2 of ICORF is set then smooth RMS and RC's, */
+    if ((icorf & bit[1]) != 0) {
+	if ((i__1 = drms[1] - drms[0], (real) abs(i__1)) >= corth[ixcor + 3] 
+		&& (i__2 = drms[1] - drms[2], (real) abs(i__2)) >= corth[
+		ixcor + 3]) {
+	    *irms = median_(&drms[2], &drms[1], drms);
+	}
+	for (i__ = 1; i__ <= 6; ++i__) {
+	    if ((i__1 = drc[i__ * 3 - 2] - drc[i__ * 3 - 3], (real) abs(i__1))
+		     >= corth[ixcor + (i__ + 2 << 2) - 5] && (i__2 = drc[i__ *
+		     3 - 2] - drc[i__ * 3 - 1], (real) abs(i__2)) >= corth[
+		    ixcor + (i__ + 2 << 2) - 5]) {
+		irc[i__] = median_(&drc[i__ * 3 - 1], &drc[i__ * 3 - 2], &drc[
+			i__ * 3 - 3]);
+	    }
+	}
+    }
+/*  If bit 3 of ICORF is set then smooth pitch */
+    if ((icorf & bit[2]) != 0) {
+	if ((i__1 = dpit[1] - dpit[0], (real) abs(i__1)) >= corth[ixcor - 1] 
+		&& (i__2 = dpit[1] - dpit[2], (real) abs(i__2)) >= corth[
+		ixcor - 1]) {
+	    *pitch = median_(&dpit[2], &dpit[1], dpit);
+	}
+    }
+/*  If bit 5 of ICORF is set then RC(5) - RC(10) are loaded with */
+/*  values so that after quantization bias is removed in decode */
+/*  the values will be zero. */
+L500:
+    if ((icorf & bit[4]) != 0) {
+	i__1 = contrl_1.order;
+	for (i__ = 5; i__ <= i__1; ++i__) {
+	    irc[i__] = zrc[i__ - 1];
+	}
+    }
+/*  House keeping  - one frame delay */
+    *iovoic = ivoic;
+    *ivp2h = voice[2];
+    dpit[2] = dpit[1];
+    dpit[1] = dpit[0];
+    drms[2] = drms[1];
+    drms[1] = drms[0];
+    i__1 = contrl_1.order;
+    for (i__ = 1; i__ <= i__1; ++i__) {
+	drc[i__ * 3 - 1] = drc[i__ * 3 - 2];
+	drc[i__ * 3 - 2] = drc[i__ * 3 - 3];
+    }
+L900:
+/* 	IF (LISTL.GE.3)WRITE(FDEBUG,801)VOICE,PITCH,IRMS,(IRC(J),J=1,ORDER) */
+/* 801	FORMAT(1X,'<<ERRCOR OUT>>',T32,2I3,I6,I5,T50,10I8) */
+/*   Decode RMS */
+    *irms = rmst[(31 - *irms) * 2];
+/*  Decode RC(1) and RC(2) from log-area-ratios */
+/*  Protect from illegal coded value (-16) caused by bit errors */
+    for (i__ = 1; i__ <= 2; ++i__) {
+	i2 = irc[i__];
+	i1 = 0;
+	if (i2 < 0) {
+	    i1 = 1;
+	    i2 = -i2;
+	    if (i2 > 15) {
+		i2 = 0;
+	    }
+	}
+	i2 = detab7[i2 * 2];
+	if (i1 == 1) {
+	    i2 = -i2;
+	}
+	ishift = 15 - nbit[i__ - 1];
+	irc[i__] = i2 * pow_ii(&c__2, &ishift);
+    }
+/*  Decode RC(3)-RC(10) to sign plus 14 bits */
+    i__1 = contrl_1.order;
+    for (i__ = 3; i__ <= i__1; ++i__) {
+	i2 = irc[i__];
+	ishift = 15 - nbit[i__ - 1];
+	i2 *= pow_ii(&c__2, &ishift);
+	i2 += qb[i__ - 3];
+	irc[i__] = i2 * descl[i__ - 3] + deadd[i__ - 3];
+    }
+/* 	IF (LISTL.GE.3) WRITE(FDEBUG,811) IRMS, (IRC(I),I=1,ORDER) */
+/* 811	FORMAT(1X,'<<DECODE OUT>>',T45,I4,1X,10I8) */
+/*  Scale RMS and RC's to reals */
+    *rms = (real) (*irms);
+    i__1 = contrl_1.order;
+    for (i__ = 1; i__ <= i__1; ++i__) {
+	rc[i__] = irc[i__] / 16384.f;
+    }
+    return 0;
+} /* decode_ */

codecs/lpc10/deemp.c

@@ -0,0 +1,142 @@
+/*
+
+$Log$
+Revision 1.4  2000/01/05 08:20:39  markster
+Version 0.1.5 from FTP
+
+Revision 1.2  2000/01/05 08:20:39  markster
+Some OSS fixes and a few lpc changes to make it actually work
+
+ * Revision 1.2  1996/08/20  20:23:46  jaf
+ * Removed all static local variables that were SAVE'd in the Fortran
+ * code, and put them in struct lpc10_decoder_state that is passed as an
+ * argument.
+ *
+ * Removed init function, since all initialization is now done in
+ * init_lpc10_decoder_state().
+ *
+ * Revision 1.1  1996/08/19  22:32:34  jaf
+ * Initial revision
+ *
+
+*/
+
+#ifdef P_R_O_T_O_T_Y_P_E_S
+extern int deemp_(real *x, integer *n, struct lpc10_decoder_state *st);
+#endif
+
+/*  -- translated by f2c (version 19951025).
+   You must link the resulting object file with the libraries:
+	-lf2c -lm   (in that order)
+*/
+
+#include "f2c.h"
+
+/* ***************************************************************** */
+
+/* 	DEEMP Version 48 */
+
+/* $Log$
+ * Revision 1.4  2000/01/05 08:20:39  markster
+ * Version 0.1.5 from FTP
+ *
+/* Revision 1.2  2000/01/05 08:20:39  markster
+/* Some OSS fixes and a few lpc changes to make it actually work
+/*
+ * Revision 1.2  1996/08/20  20:23:46  jaf
+ * Removed all static local variables that were SAVE'd in the Fortran
+ * code, and put them in struct lpc10_decoder_state that is passed as an
+ * argument.
+ *
+ * Removed init function, since all initialization is now done in
+ * init_lpc10_decoder_state().
+ *
+ * Revision 1.1  1996/08/19  22:32:34  jaf
+ * Initial revision
+ * */
+/* Revision 1.3  1996/03/20  15:54:37  jaf */
+/* Added comments about which indices of array arguments are read or */
+/* written. */
+
+/* Added entry INITDEEMP to reinitialize the local state variables, if */
+/* desired. */
+
+/* Revision 1.2  1996/03/14  22:11:13  jaf */
+/* Comments added explaining which of the local variables of this */
+/* subroutine need to be saved from one invocation to the next, and which */
+/* do not. */
+
+/* Revision 1.1  1996/02/07 14:44:53  jaf */
+/* Initial revision */
+
+
+/* ***************************************************************** */
+
+/*  De-Emphasize output speech with   1 / ( 1 - .75z**-1 ) */
+/*    cascaded with 200 Hz high pass filter */
+/*    ( 1 - 1.9998z**-1 + z**-2 ) / ( 1 - 1.75z**-1 + .78z**-2 ) */
+
+/*  WARNING!  The coefficients above may be out of date with the code */
+/*  below.  Either that, or some kind of transformation was performed */
+/*  on the coefficients above to create the code below. */
+
+/* Input: */
+/*  N  - Number of samples */
+/* Input/Output: */
+/*  X  - Speech */
+/*       Indices 1 through N are read before being written. */
+
+/* This subroutine maintains local state from one call to the next.  If */
+/* you want to switch to using a new audio stream for this filter, or */
+/* reinitialize its state for any other reason, call the ENTRY */
+/* INITDEEMP. */
+
+/* Subroutine */ int deemp_(real *x, integer *n, struct lpc10_decoder_state *st)
+{
+    /* Initialized data */
+
+    real *dei1;
+    real *dei2;
+    real *deo1;
+    real *deo2;
+    real *deo3;
+
+    /* System generated locals */
+    integer i__1;
+    real r__1;
+
+    /* Local variables */
+    integer k;
+    real dei0;
+
+/*       Arguments */
+/*       Local variables that need not be saved */
+/*       Local state */
+/*       All of the locals saved below were not given explicit initial */
+/*       values in the original code.  I think 0 is a safe choice. */
+    /* Parameter adjustments */
+    if (x) {
+	--x;
+	}
+
+    /* Function Body */
+
+    dei1 = &(st->dei1);
+    dei2 = &(st->dei2);
+    deo1 = &(st->deo1);
+    deo2 = &(st->deo2);
+    deo3 = &(st->deo3);
+
+    i__1 = *n;
+    for (k = 1; k <= i__1; ++k) {
+	dei0 = x[k];
+	r__1 = x[k] - *dei1 * 1.9998f + *dei2;
+	x[k] = r__1 + *deo1 * 2.5f - *deo2 * 2.0925f + *deo3 * .585f;
+	*dei2 = *dei1;
+	*dei1 = dei0;
+	*deo3 = *deo2;
+	*deo2 = *deo1;
+	*deo1 = x[k];
+    }
+    return 0;
+} /* deemp_ */

codecs/lpc10/difmag.c

@@ -0,0 +1,121 @@
+/*
+
+$Log$
+Revision 1.4  2000/01/05 08:20:39  markster
+Version 0.1.5 from FTP
+
+Revision 1.2  2000/01/05 08:20:39  markster
+Some OSS fixes and a few lpc changes to make it actually work
+
+ * Revision 1.1  1996/08/19  22:32:31  jaf
+ * Initial revision
+ *
+
+*/
+
+#ifdef P_R_O_T_O_T_Y_P_E_S
+extern int difmag_(real *speech, integer *lpita, integer *tau, integer *ltau, integer *maxlag, real *amdf, integer *minptr, integer *maxptr);
+#endif
+
+/*  -- translated by f2c (version 19951025).
+   You must link the resulting object file with the libraries:
+	-lf2c -lm   (in that order)
+*/
+
+#include "f2c.h"
+
+/* ********************************************************************** */
+
+/* 	DIFMAG Version 49 */
+
+/* $Log$
+ * Revision 1.4  2000/01/05 08:20:39  markster
+ * Version 0.1.5 from FTP
+ *
+/* Revision 1.2  2000/01/05 08:20:39  markster
+/* Some OSS fixes and a few lpc changes to make it actually work
+/*
+ * Revision 1.1  1996/08/19  22:32:31  jaf
+ * Initial revision
+ * */
+/* Revision 1.3  1996/03/15  23:09:39  jaf */
+/* Just added a few comments about which array indices of the arguments */
+/* are used, and mentioning that this subroutine has no local state. */
+
+/* Revision 1.2  1996/03/13  14:41:31  jaf */
+/* Comments added explaining that none of the local variables of this */
+/* subroutine need to be saved from one invocation to the next. */
+
+/* Revision 1.1  1996/02/07 14:45:04  jaf */
+/* Initial revision */
+
+
+/* ********************************************************************* */
+
+/*  Compute Average Magnitude Difference Function */
+
+/* Inputs: */
+/*  SPEECH - Low pass filtered speech */
+/*           Indices MIN_N1 through MAX_N1+LPITA-1 are read, where */
+/*      MIN_N1 = (MAXLAG - MAX_TAU)/2+1  MAX_TAU = max of TAU(I) for I=1,LTAU
+*/
+/*      MAX_N1 = (MAXLAG - MIN_TAU)/2+1  MIN_TAU = min of TAU(I) for I=1,LTAU
+*/
+/*  LPITA  - Length of speech buffer */
+/*  TAU    - Table of lags */
+/*           Indices 1 through LTAU read. */
+/*  LTAU   - Number of lag values to compute */
+/*  MAXLAG - Maximum possible lag value */
+/* Outputs: */
+/*  (All of these outputs are also read, but only after being written.) */
+/*  AMDF   - Average Magnitude Difference for each lag in TAU */
+/*           Indices 1 through LTAU written */
+/*  MINPTR - Index of minimum AMDF value */
+/*  MAXPTR - Index of maximum AMDF value */
+
+/* This subroutine has no local state. */
+
+/* Subroutine */ int difmag_(real *speech, integer *lpita, integer *tau, 
+	integer *ltau, integer *maxlag, real *amdf, integer *minptr, integer *
+	maxptr)
+{
+    /* System generated locals */
+    integer i__1, i__2;
+    real r__1;
+
+    /* Local variables */
+    integer i__, j, n1, n2;
+    real sum;
+
+/*       Arguments */
+/*       Local variables that need not be saved */
+/*       Local state */
+/*       None */
+    /* Parameter adjustments */
+    --amdf;
+    --tau;
+    --speech;
+
+    /* Function Body */
+    *minptr = 1;
+    *maxptr = 1;
+    i__1 = *ltau;
+    for (i__ = 1; i__ <= i__1; ++i__) {
+	n1 = (*maxlag - tau[i__]) / 2 + 1;
+	n2 = n1 + *lpita - 1;
+	sum = 0.f;
+	i__2 = n2;
+	for (j = n1; j <= i__2; j += 4) {
+	    sum += (r__1 = speech[j] - speech[j + tau[i__]], abs(r__1));
+	}
+	amdf[i__] = sum;
+	if (amdf[i__] < amdf[*minptr]) {
+	    *minptr = i__;
+	}
+	if (amdf[i__] > amdf[*maxptr]) {
+	    *maxptr = i__;
+	}
+    }
+    return 0;
+} /* difmag_ */
+

codecs/lpc10/dyptrk.c

@@ -0,0 +1,387 @@
+/*
+
+$Log$
+Revision 1.4  2000/01/05 08:20:39  markster
+Version 0.1.5 from FTP
+
+Revision 1.2  2000/01/05 08:20:39  markster
+Some OSS fixes and a few lpc changes to make it actually work
+
+ * Revision 1.2  1996/08/20  20:25:29  jaf
+ * Removed all static local variables that were SAVE'd in the Fortran
+ * code, and put them in struct lpc10_encoder_state that is passed as an
+ * argument.
+ *
+ * Removed init function, since all initialization is now done in
+ * init_lpc10_encoder_state().
+ *
+ * Revision 1.1  1996/08/19  22:32:26  jaf
+ * Initial revision
+ *
+
+*/
+
+#ifdef P_R_O_T_O_T_Y_P_E_S
+extern int dyptrk_(real *amdf, integer *ltau, integer *minptr, integer *voice, integer *pitch, integer *midx, struct lpc10_encoder_state *st);
+/* comlen contrl_ 12 */
+#endif
+
+/*  -- translated by f2c (version 19951025).
+   You must link the resulting object file with the libraries:
+	-lf2c -lm   (in that order)
+*/
+
+#include "f2c.h"
+
+/* Common Block Declarations */
+
+extern struct {
+    integer order, lframe;
+    logical corrp;
+} contrl_;
+
+#define contrl_1 contrl_
+
+/* ********************************************************************* */
+
+/* 	DYPTRK Version 52 */
+
+/* $Log$
+ * Revision 1.4  2000/01/05 08:20:39  markster
+ * Version 0.1.5 from FTP
+ *
+/* Revision 1.2  2000/01/05 08:20:39  markster
+/* Some OSS fixes and a few lpc changes to make it actually work
+/*
+ * Revision 1.2  1996/08/20  20:25:29  jaf
+ * Removed all static local variables that were SAVE'd in the Fortran
+ * code, and put them in struct lpc10_encoder_state that is passed as an
+ * argument.
+ *
+ * Removed init function, since all initialization is now done in
+ * init_lpc10_encoder_state().
+ *
+ * Revision 1.1  1996/08/19  22:32:26  jaf
+ * Initial revision
+ * */
+/* Revision 1.5  1996/03/26  19:35:35  jaf */
+/* Commented out trace statements. */
+
+/* Revision 1.4  1996/03/19  18:03:22  jaf */
+/* Replaced the initialization "DATA P/60*DEPTH*0/" with "DATA P/120*0/", */
+/* because apparently Fortran (or at least f2c) can't handle expressions */
+/* like that. */
+
+/* Revision 1.3  1996/03/19  17:38:32  jaf */
+/* Added comments about the local variables that should be saved from one */
+/* invocation to the next.  None of them were given initial values in the */
+/* original code, but from my testing, it appears that initializing them */
+/* all to 0 works. */
+
+/* Added entry INITDYPTRK to reinitialize these local variables. */
+
+/* Revision 1.2  1996/03/13  16:32:17  jaf */
+/* Comments added explaining which of the local variables of this */
+/* subroutine need to be saved from one invocation to the next, and which */
+/* do not. */
+
+/* WARNING!  Some of them that should are never given initial values in */
+/* this code.  Hopefully, Fortran 77 defines initial values for them, but */
+/* even so, giving them explicit initial values is preferable. */
+
+/* Revision 1.1  1996/02/07 14:45:14  jaf */
+/* Initial revision */
+
+
+/* ********************************************************************* */
+
+/*   Dynamic Pitch Tracker */
+
+/* Input: */
+/*  AMDF   - Average Magnitude Difference Function array */
+/*           Indices 1 through LTAU read, and MINPTR */
+/*  LTAU   - Number of lags in AMDF */
+/*  MINPTR - Location of minimum AMDF value */
+/*  VOICE  - Voicing decision */
+/* Output: */
+/*  PITCH  - Smoothed pitch value, 2 frames delayed */
+/*  MIDX   - Initial estimate of current frame pitch */
+/* Compile time constant: */
+/*  DEPTH  - Number of frames to trace back */
+
+/* This subroutine maintains local state from one call to the next.  If */
+/* you want to switch to using a new audio stream for this filter, or */
+/* reinitialize its state for any other reason, call the ENTRY */
+/* INITDYPTRK. */
+
+/* Subroutine */ int dyptrk_(real *amdf, integer *ltau, integer *
+	minptr, integer *voice, integer *pitch, integer *midx,
+			       struct lpc10_encoder_state *st)
+{
+    /* Initialized data */
+
+    real *s;
+    integer *p;
+    integer *ipoint;
+    real *alphax;
+
+    /* System generated locals */
+    integer i__1;
+
+    /* Local variables */
+    integer pbar;
+    real sbar;
+    integer path[2], iptr, i__, j;
+    real alpha, minsc, maxsc;
+
+/*       Arguments */
+/* $Log$
+ * Revision 1.4  2000/01/05 08:20:39  markster
+ * Version 0.1.5 from FTP
+ *
+/* Revision 1.2  2000/01/05 08:20:39  markster
+/* Some OSS fixes and a few lpc changes to make it actually work
+/*
+ * Revision 1.2  1996/08/20  20:25:29  jaf
+ * Removed all static local variables that were SAVE'd in the Fortran
+ * code, and put them in struct lpc10_encoder_state that is passed as an
+ * argument.
+ *
+ * Removed init function, since all initialization is now done in
+ * init_lpc10_encoder_state().
+ *
+ * Revision 1.1  1996/08/19  22:32:26  jaf
+ * Initial revision
+ * */
+/* Revision 1.3  1996/03/29  22:05:55  jaf */
+/* Commented out the common block variables that are not needed by the */
+/* embedded version. */
+
+/* Revision 1.2  1996/03/26  19:34:50  jaf */
+/* Added comments indicating which constants are not needed in an */
+/* application that uses the LPC-10 coder. */
+
+/* Revision 1.1  1996/02/07  14:44:09  jaf */
+/* Initial revision */
+
+/*   LPC Processing control variables: */
+
+/* *** Read-only: initialized in setup */
+
+/*  Files for Speech, Parameter, and Bitstream Input & Output, */
+/*    and message and debug outputs. */
+
+/* Here are the only files which use these variables: */
+
+/* lpcsim.f setup.f trans.f error.f vqsetup.f */
+
+/* Many files which use fdebug are not listed, since it is only used in */
+/* those other files conditionally, to print trace statements. */
+/* 	integer fsi, fso, fpi, fpo, fbi, fbo, pbin, fmsg, fdebug */
+/*  LPC order, Frame size, Quantization rate, Bits per frame, */
+/*    Error correction */
+/* Subroutine SETUP is the only place where order is assigned a value, */
+/* and that value is 10.  It could increase efficiency 1% or so to */
+/* declare order as a constant (i.e., a Fortran PARAMETER) instead of as 
+*/
+/* a variable in a COMMON block, since it is used in many places in the */
+/* core of the coding and decoding routines.  Actually, I take that back. 
+*/
+/* At least when compiling with f2c, the upper bound of DO loops is */
+/* stored in a local variable before the DO loop begins, and then that is 
+*/
+/* compared against on each iteration. */
+/* Similarly for lframe, which is given a value of MAXFRM in SETUP. */
+/* Similarly for quant, which is given a value of 2400 in SETUP.  quant */
+/* is used in only a few places, and never in the core coding and */
+/* decoding routines, so it could be eliminated entirely. */
+/* nbits is similar to quant, and is given a value of 54 in SETUP. */
+/* corrp is given a value of .TRUE. in SETUP, and is only used in the */
+/* subroutines ENCODE and DECODE.  It doesn't affect the speed of the */
+/* coder significantly whether it is .TRUE. or .FALSE., or whether it is 
+*/
+/* a constant or a variable, since it is only examined once per frame. */
+/* Leaving it as a variable that is set to .TRUE.  seems like a good */
+/* idea, since it does enable some error-correction capability for */
+/* unvoiced frames, with no change in the coding rate, and no noticeable 
+*/
+/* quality difference in the decoded speech. */
+/* 	integer quant, nbits */
+/* *** Read/write: variables for debugging, not needed for LPC algorithm 
+*/
+
+/*  Current frame, Unstable frames, Output clip count, Max onset buffer, 
+*/
+/*    Debug listing detail level, Line count on listing page */
+
+/* nframe is not needed for an embedded LPC10 at all. */
+/* nunsfm is initialized to 0 in SETUP, and incremented in subroutine */
+/* ERROR, which is only called from RCCHK.  When LPC10 is embedded into */
+/* an application, I would recommend removing the call to ERROR in RCCHK, 
+*/
+/* and remove ERROR and nunsfm completely. */
+/* iclip is initialized to 0 in SETUP, and incremented in entry SWRITE in 
+*/
+/* sread.f.  When LPC10 is embedded into an application, one might want */
+/* to cause it to be incremented in a routine that takes the output of */
+/* SYNTHS and sends it to an audio device.  It could be optionally */
+/* displayed, for those that might want to know what it is. */
+/* maxosp is never initialized to 0 in SETUP, although it probably should 
+*/
+/* be, and it is updated in subroutine ANALYS.  I doubt that its value */
+/* would be of much interest to an application in which LPC10 is */
+/* embedded. */
+/* listl and lincnt are not needed for an embedded LPC10 at all. */
+/* 	integer nframe, nunsfm, iclip, maxosp, listl, lincnt */
+/* 	common /contrl/ fsi, fso, fpi, fpo, fbi, fbo, pbin, fmsg, fdebug */
+/* 	common /contrl/ quant, nbits */
+/* 	common /contrl/ nframe, nunsfm, iclip, maxosp, listl, lincnt */
+/* 	Parameters/constants */
+/*       Local variables that need not be saved */
+/*       Note that PATH is only used for debugging purposes, and can be */
+/*       removed. */
+/*       Local state */
+/*       It would be a bit more "general" to define S(LTAU), if Fortran */
+/*       allows the argument of a function to be used as the dimension of 
+*/
+/*       a local array variable. */
+/*       IPOINT is always in the range 0 to DEPTH-1. */
+/*       WARNING! */
+
+/*       In the original version of this subroutine, IPOINT, ALPHAX, */
+/*       every element of S, and potentially any element of P with the */
+/*       second index value .NE. IPTR were read without being given */
+/*       initial values (all indices of P with second index equal to */
+/*       IPTR are all written before being read in this subroutine). */
+
+/*       From examining the code carefully, it appears that all of these 
+*/
+/*       should be saved from one invocation to the next. */
+
+/*       I've run lpcsim with the "-l 6" option to see all of the */
+/*       debugging information that is printed out by this subroutine */
+/*       below, and it appears that S, P, IPOINT, and ALPHAX are all */
+/*       initialized to 0 (these initial values would likely be different 
+*/
+/*       on different platforms, compilers, etc.).  Given that the output 
+*/
+/*       of the coder sounds reasonable, I'm going to initialize these */
+/*       variables to 0 explicitly. */
+
+    s = &(st->s[0]);
+    p = &(st->p[0]);
+    ipoint = &(st->ipoint);
+    alphax = &(st->alphax);
+
+
+    /* Parameter adjustments */
+    if (amdf) {
+	--amdf;
+	}
+
+    /* Function Body */
+
+/*   Calculate the confidence factor ALPHA, used as a threshold slope in 
+*/
+/*   SEESAW.  If unvoiced, set high slope so that every point in P array 
+*/
+/*  is marked as a potential pitch frequency.  A scaled up version (ALPHAX
+)*/
+/*   is used to maintain arithmetic precision. */
+    if (*voice == 1) {
+	*alphax = *alphax * .75f + amdf[*minptr] / 2.f;
+    } else {
+	*alphax *= .984375f;
+    }
+    alpha = *alphax / 16;
+    if (*voice == 0 && *alphax < 128.f) {
+	alpha = 8.f;
+    }
+/* SEESAW: Construct a pitch pointer array and intermediate winner functio
+n*/
+/*   Left to right pass: */
+    iptr = *ipoint + 1;
+    p[iptr * 60 - 60] = 1;
+    i__ = 1;
+    pbar = 1;
+    sbar = s[0];
+    i__1 = *ltau;
+    for (i__ = 1; i__ <= i__1; ++i__) {
+	sbar += alpha;
+	if (sbar < s[i__ - 1]) {
+	    s[i__ - 1] = sbar;
+	    p[i__ + iptr * 60 - 61] = pbar;
+	} else {
+	    sbar = s[i__ - 1];
+	    p[i__ + iptr * 60 - 61] = i__;
+	    pbar = i__;
+	}
+    }
+/*   Right to left pass: */
+    i__ = pbar - 1;
+    sbar = s[i__];
+    while(i__ >= 1) {
+	sbar += alpha;
+	if (sbar < s[i__ - 1]) {
+	    s[i__ - 1] = sbar;
+	    p[i__ + iptr * 60 - 61] = pbar;
+	} else {
+	    pbar = p[i__ + iptr * 60 - 61];
+	    i__ = pbar;
+	    sbar = s[i__ - 1];
+	}
+	--i__;
+    }
+/*   Update S using AMDF */
+/*   Find maximum, minimum, and location of minimum */
+    s[0] += amdf[1] / 2;
+    minsc = s[0];
+    maxsc = minsc;
+    *midx = 1;
+    i__1 = *ltau;
+    for (i__ = 2; i__ <= i__1; ++i__) {
+	s[i__ - 1] += amdf[i__] / 2;
+	if (s[i__ - 1] > maxsc) {
+	    maxsc = s[i__ - 1];
+	}
+	if (s[i__ - 1] < minsc) {
+	    *midx = i__;
+	    minsc = s[i__ - 1];
+	}
+    }
+/*   Subtract MINSC from S to prevent overflow */
+    i__1 = *ltau;
+    for (i__ = 1; i__ <= i__1; ++i__) {
+	s[i__ - 1] -= minsc;
+    }
+    maxsc -= minsc;
+/*   Use higher octave pitch if significant null there */
+    j = 0;
+    for (i__ = 20; i__ <= 40; i__ += 10) {
+	if (*midx > i__) {
+	    if (s[*midx - i__ - 1] < maxsc / 4) {
+		j = i__;
+	    }
+	}
+    }
+    *midx -= j;
+/*   TRACE: look back two frames to find minimum cost pitch estimate */
+    j = *ipoint;
+    *pitch = *midx;
+    for (i__ = 1; i__ <= 2; ++i__) {
+	j = j % 2 + 1;
+	*pitch = p[*pitch + j * 60 - 61];
+	path[i__ - 1] = *pitch;
+    }
+
+/*       The following statement subtracts one from IPOINT, mod DEPTH.  I 
+*/
+/*       think the author chose to add DEPTH-1, instead of subtracting 1, 
+*/
+/*       because then it will work even if MOD doesn't work as desired on 
+*/
+/*       negative arguments. */
+
+    *ipoint = (*ipoint + 1) % 2;
+    return 0;
+} /* dyptrk_ */

codecs/lpc10/encode.c

@@ -0,0 +1,349 @@
+/*
+
+$Log$
+Revision 1.4  2000/01/05 08:20:39  markster
+Version 0.1.5 from FTP
+
+Revision 1.2  2000/01/05 08:20:39  markster
+Some OSS fixes and a few lpc changes to make it actually work
+
+ * Revision 1.1  1996/08/19  22:32:21  jaf
+ * Initial revision
+ *
+
+*/
+
+#ifdef P_R_O_T_O_T_Y_P_E_S
+extern int encode_(integer *voice, integer *pitch, real *rms, real *rc, integer *ipitch, integer *irms, integer *irc);
+/* comlen contrl_ 12 */
+#endif
+
+/*  -- translated by f2c (version 19951025).
+   You must link the resulting object file with the libraries:
+	-lf2c -lm   (in that order)
+*/
+
+#include "f2c.h"
+
+/* Common Block Declarations */
+
+extern struct {
+    integer order, lframe;
+    logical corrp;
+} contrl_;
+
+#define contrl_1 contrl_
+
+/* Table of constant values */
+
+static integer c__2 = 2;
+
+/* ***************************************************************** */
+
+/* 	ENCODE Version 54 */
+
+/* $Log$
+ * Revision 1.4  2000/01/05 08:20:39  markster
+ * Version 0.1.5 from FTP
+ *
+/* Revision 1.2  2000/01/05 08:20:39  markster
+/* Some OSS fixes and a few lpc changes to make it actually work
+/*
+ * Revision 1.1  1996/08/19  22:32:21  jaf
+ * Initial revision
+ * */
+/* Revision 1.5  1996/03/26  19:35:50  jaf */
+/* Commented out trace statements. */
+
+/* Revision 1.4  1996/03/21  00:26:29  jaf */
+/* Added the comment that this subroutine has no local state. */
+
+/* In the last check-in, I forgot to mention that I had added comments */
+/* explaining which indices of array arguments are read or written. */
+
+/* Revision 1.3  1996/03/21  00:22:39  jaf */
+/* Added comments explaining that all local arrays are effectively */
+/* constants. */
+
+/* Revision 1.2  1996/03/13  18:48:33  jaf */
+/* Comments added explaining that none of the local variables of this */
+/* subroutine need to be saved from one invocation to the next. */
+
+/* Revision 1.1  1996/02/07 14:45:29  jaf */
+/* Initial revision */
+
+
+/* ***************************************************************** */
+
+/*  Quantize LPC parameters for transmission */
+
+/* INPUTS: */
+/*  VOICE  - Half frame voicing decisions */
+/*           Indices 1 through 2 read. */
+/*  PITCH  - Pitch */
+/*  RMS    - Energy */
+/*  RC     - Reflection coefficients */
+/*           Indices 1 through ORDER read. */
+/*  CORRP  - Error Correction: TRUE = yes, FALSE = none */
+/*           (this is defined in file control.fh) */
+/* OUTPUTS: */
+/*  IPITCH - Coded pitch and voicing */
+/*  IRMS   - Quantized energy */
+/*  IRC    - Quantized reflection coefficients */
+/*           Indices 1 through MAX(ORDER,2) written. */
+/*           If CORRP is .TRUE., then indices 1 through 10 written */
+/*           for unvoiced frames. */
+
+/* This subroutine has no local state. */
+
+/* Subroutine */ int encode_(integer *voice, integer *pitch, real *rms, real *
+	rc, integer *ipitch, integer *irms, integer *irc)
+{
+    /* Initialized data */
+
+    static integer enctab[16] = { 0,7,11,12,13,10,6,1,14,9,5,2,3,4,8,15 };
+    static integer entau[60] = { 19,11,27,25,29,21,23,22,30,14,15,7,39,38,46,
+	    42,43,41,45,37,53,49,51,50,54,52,60,56,58,26,90,88,92,84,86,82,83,
+	    81,85,69,77,73,75,74,78,70,71,67,99,97,113,112,114,98,106,104,108,
+	    100,101,76 };
+    static integer enadd[8] = { 1920,-768,2432,1280,3584,1536,2816,-1152 };
+    static real enscl[8] = { .0204f,.0167f,.0145f,.0147f,.0143f,.0135f,.0125f,
+	    .0112f };
+    static integer enbits[8] = { 6,5,4,4,4,4,3,3 };
+    static integer entab6[64] = { 0,0,0,0,0,0,1,1,1,1,1,1,1,2,2,2,2,2,2,2,3,3,
+	    3,3,3,3,3,4,4,4,4,4,4,4,5,5,5,5,5,6,6,6,6,6,7,7,7,7,7,8,8,8,8,9,9,
+	    9,10,10,11,11,12,13,14,15 };
+    static integer rmst[64] = { 1024,936,856,784,718,656,600,550,502,460,420,
+	    384,352,328,294,270,246,226,206,188,172,158,144,132,120,110,102,
+	    92,84,78,70,64,60,54,50,46,42,38,34,32,30,26,24,22,20,18,17,16,15,
+	    14,13,12,11,10,9,8,7,6,5,4,3,2,1,0 };
+
+    /* System generated locals */
+    integer i__1, i__2;
+
+    /* Builtin functions */
+    integer pow_ii(integer *, integer *);
+
+    /* Local variables */
+    integer idel, nbit, i__, j, i2, i3, mrk;
+
+/* $Log$
+ * Revision 1.4  2000/01/05 08:20:39  markster
+ * Version 0.1.5 from FTP
+ *
+/* Revision 1.2  2000/01/05 08:20:39  markster
+/* Some OSS fixes and a few lpc changes to make it actually work
+/*
+ * Revision 1.1  1996/08/19  22:32:21  jaf
+ * Initial revision
+ * */
+/* Revision 1.3  1996/03/29  22:03:47  jaf */
+/* Removed definitions for any constants that were no longer used. */
+
+/* Revision 1.2  1996/03/26  19:34:33  jaf */
+/* Added comments indicating which constants are not needed in an */
+/* application that uses the LPC-10 coder. */
+
+/* Revision 1.1  1996/02/07  14:43:51  jaf */
+/* Initial revision */
+
+/*   LPC Configuration parameters: */
+/* Frame size, Prediction order, Pitch period */
+/*       Arguments */
+/* $Log$
+ * Revision 1.4  2000/01/05 08:20:39  markster
+ * Version 0.1.5 from FTP
+ *
+/* Revision 1.2  2000/01/05 08:20:39  markster
+/* Some OSS fixes and a few lpc changes to make it actually work
+/*
+ * Revision 1.1  1996/08/19  22:32:21  jaf
+ * Initial revision
+ * */
+/* Revision 1.3  1996/03/29  22:05:55  jaf */
+/* Commented out the common block variables that are not needed by the */
+/* embedded version. */
+
+/* Revision 1.2  1996/03/26  19:34:50  jaf */
+/* Added comments indicating which constants are not needed in an */
+/* application that uses the LPC-10 coder. */
+
+/* Revision 1.1  1996/02/07  14:44:09  jaf */
+/* Initial revision */
+
+/*   LPC Processing control variables: */
+
+/* *** Read-only: initialized in setup */
+
+/*  Files for Speech, Parameter, and Bitstream Input & Output, */
+/*    and message and debug outputs. */
+
+/* Here are the only files which use these variables: */
+
+/* lpcsim.f setup.f trans.f error.f vqsetup.f */
+
+/* Many files which use fdebug are not listed, since it is only used in */
+/* those other files conditionally, to print trace statements. */
+/* 	integer fsi, fso, fpi, fpo, fbi, fbo, pbin, fmsg, fdebug */
+/*  LPC order, Frame size, Quantization rate, Bits per frame, */
+/*    Error correction */
+/* Subroutine SETUP is the only place where order is assigned a value, */
+/* and that value is 10.  It could increase efficiency 1% or so to */
+/* declare order as a constant (i.e., a Fortran PARAMETER) instead of as 
+*/
+/* a variable in a COMMON block, since it is used in many places in the */
+/* core of the coding and decoding routines.  Actually, I take that back. 
+*/
+/* At least when compiling with f2c, the upper bound of DO loops is */
+/* stored in a local variable before the DO loop begins, and then that is 
+*/
+/* compared against on each iteration. */
+/* Similarly for lframe, which is given a value of MAXFRM in SETUP. */
+/* Similarly for quant, which is given a value of 2400 in SETUP.  quant */
+/* is used in only a few places, and never in the core coding and */
+/* decoding routines, so it could be eliminated entirely. */
+/* nbits is similar to quant, and is given a value of 54 in SETUP. */
+/* corrp is given a value of .TRUE. in SETUP, and is only used in the */
+/* subroutines ENCODE and DECODE.  It doesn't affect the speed of the */
+/* coder significantly whether it is .TRUE. or .FALSE., or whether it is 
+*/
+/* a constant or a variable, since it is only examined once per frame. */
+/* Leaving it as a variable that is set to .TRUE.  seems like a good */
+/* idea, since it does enable some error-correction capability for */
+/* unvoiced frames, with no change in the coding rate, and no noticeable 
+*/
+/* quality difference in the decoded speech. */
+/* 	integer quant, nbits */
+/* *** Read/write: variables for debugging, not needed for LPC algorithm 
+*/
+
+/*  Current frame, Unstable frames, Output clip count, Max onset buffer, 
+*/
+/*    Debug listing detail level, Line count on listing page */
+
+/* nframe is not needed for an embedded LPC10 at all. */
+/* nunsfm is initialized to 0 in SETUP, and incremented in subroutine */
+/* ERROR, which is only called from RCCHK.  When LPC10 is embedded into */
+/* an application, I would recommend removing the call to ERROR in RCCHK, 
+*/
+/* and remove ERROR and nunsfm completely. */
+/* iclip is initialized to 0 in SETUP, and incremented in entry SWRITE in 
+*/
+/* sread.f.  When LPC10 is embedded into an application, one might want */
+/* to cause it to be incremented in a routine that takes the output of */
+/* SYNTHS and sends it to an audio device.  It could be optionally */
+/* displayed, for those that might want to know what it is. */
+/* maxosp is never initialized to 0 in SETUP, although it probably should 
+*/
+/* be, and it is updated in subroutine ANALYS.  I doubt that its value */
+/* would be of much interest to an application in which LPC10 is */
+/* embedded. */
+/* listl and lincnt are not needed for an embedded LPC10 at all. */
+/* 	integer nframe, nunsfm, iclip, maxosp, listl, lincnt */
+/* 	common /contrl/ fsi, fso, fpi, fpo, fbi, fbo, pbin, fmsg, fdebug */
+/* 	common /contrl/ quant, nbits */
+/* 	common /contrl/ nframe, nunsfm, iclip, maxosp, listl, lincnt */
+/*       Parameters/constants */
+/*       These arrays are not Fortran PARAMETER's, but they are defined */
+/*       by DATA statements below, and their contents are never altered. 
+*/
+/*       Local variables that need not be saved */
+    /* Parameter adjustments */
+    --irc;
+    --rc;
+    --voice;
+
+    /* Function Body */
+/*  Scale RMS and RC's to integers */
+    *irms = *rms;
+    i__1 = contrl_1.order;
+    for (i__ = 1; i__ <= i__1; ++i__) {
+	irc[i__] = rc[i__] * 32768.f;
+    }
+/* 	IF(LISTL.GE.3)WRITE(FDEBUG,800)VOICE,PITCH,IRMS,(IRC(I),I=1,ORDER) */
+/* 800	FORMAT(1X,/,' <<ENCODE IN>>',T32,2I3,I6,I5,T50,10I8) */
+/*  Encode pitch and voicing */
+    if (voice[1] != 0 && voice[2] != 0) {
+	*ipitch = entau[*pitch - 1];
+    } else {
+	if (contrl_1.corrp) {
+	    *ipitch = 0;
+	    if (voice[1] != voice[2]) {
+		*ipitch = 127;
+	    }
+	} else {
+	    *ipitch = (voice[1] << 1) + voice[2];
+	}
+    }
+/*  Encode RMS by binary table search */
+    j = 32;
+    idel = 16;
+    *irms = min(*irms,1023);
+    while(idel > 0) {
+	if (*irms > rmst[j - 1]) {
+	    j -= idel;
+	}
+	if (*irms < rmst[j - 1]) {
+	    j += idel;
+	}
+	idel /= 2;
+    }
+    if (*irms > rmst[j - 1]) {
+	--j;
+    }
+    *irms = 31 - j / 2;
+/*  Encode RC(1) and (2) as log-area-ratios */
+    for (i__ = 1; i__ <= 2; ++i__) {
+	i2 = irc[i__];
+	mrk = 0;
+	if (i2 < 0) {
+	    i2 = -i2;
+	    mrk = 1;
+	}
+	i2 /= 512;
+	i2 = min(i2,63);
+	i2 = entab6[i2];
+	if (mrk != 0) {
+	    i2 = -i2;
+	}
+	irc[i__] = i2;
+    }
+/*  Encode RC(3) - (10) linearly, remove bias then scale */
+    i__1 = contrl_1.order;
+    for (i__ = 3; i__ <= i__1; ++i__) {
+	i2 = irc[i__] / 2;
+	i2 = (i2 + enadd[contrl_1.order + 1 - i__ - 1]) * enscl[
+		contrl_1.order + 1 - i__ - 1];
+/* Computing MIN */
+	i__2 = max(i2,-127);
+	i2 = min(i__2,127);
+	nbit = enbits[contrl_1.order + 1 - i__ - 1];
+	i3 = 0;
+	if (i2 < 0) {
+	    i3 = -1;
+	}
+	i2 /= pow_ii(&c__2, &nbit);
+	if (i3 == -1) {
+	    --i2;
+	}
+	irc[i__] = i2;
+    }
+/*          Protect the most significant bits of the most */
+/*     important parameters during non-voiced frames. */
+/*     RC(1) - RC(4) are protected using 20 parity bits */
+/*     replacing RC(5) - RC(10). */
+    if (contrl_1.corrp) {
+	if (*ipitch == 0 || *ipitch == 127) {
+	    irc[5] = enctab[(irc[1] & 30) / 2];
+	    irc[6] = enctab[(irc[2] & 30) / 2];
+	    irc[7] = enctab[(irc[3] & 30) / 2];
+	    irc[8] = enctab[(*irms & 30) / 2];
+	    irc[9] = enctab[(irc[4] & 30) / 2] / 2;
+	    irc[10] = enctab[(irc[4] & 30) / 2] & 1;
+	}
+    }
+/* 	IF(LISTL.GE.3)WRITE(FDEBUG,801)VOICE,IPITCH,IRMS,(IRC(J),J=1,ORDER) */
+/* 801	FORMAT(1X,'<<ENCODE OUT>>',T32,2I3,I6,I5,T50,10I8) */
+    return 0;
+} /* encode_ */
+

codecs/lpc10/energy.c

@@ -0,0 +1,91 @@
+/*
+
+$Log$
+Revision 1.4  2000/01/05 08:20:39  markster
+Version 0.1.5 from FTP
+
+Revision 1.2  2000/01/05 08:20:39  markster
+Some OSS fixes and a few lpc changes to make it actually work
+
+ * Revision 1.1  1996/08/19  22:32:17  jaf
+ * Initial revision
+ *
+
+*/
+
+#ifdef P_R_O_T_O_T_Y_P_E_S
+extern int energy_(integer *len, real *speech, real *rms);
+#endif
+
+/*  -- translated by f2c (version 19951025).
+   You must link the resulting object file with the libraries:
+	-lf2c -lm   (in that order)
+*/
+
+#include "f2c.h"
+
+/* ********************************************************************* */
+
+/* 	ENERGY Version 50 */
+
+/* $Log$
+ * Revision 1.4  2000/01/05 08:20:39  markster
+ * Version 0.1.5 from FTP
+ *
+/* Revision 1.2  2000/01/05 08:20:39  markster
+/* Some OSS fixes and a few lpc changes to make it actually work
+/*
+ * Revision 1.1  1996/08/19  22:32:17  jaf
+ * Initial revision
+ * */
+/* Revision 1.3  1996/03/18  21:17:41  jaf */
+/* Just added a few comments about which array indices of the arguments */
+/* are used, and mentioning that this subroutine has no local state. */
+
+/* Revision 1.2  1996/03/13  16:46:02  jaf */
+/* Comments added explaining that none of the local variables of this */
+/* subroutine need to be saved from one invocation to the next. */
+
+/* Revision 1.1  1996/02/07 14:45:40  jaf */
+/* Initial revision */
+
+
+/* ********************************************************************* */
+
+/* Compute RMS energy. */
+
+/* Input: */
+/*  LEN    - Length of speech buffer */
+/*  SPEECH - Speech buffer */
+/*           Indices 1 through LEN read. */
+/* Output: */
+/*  RMS    - Root Mean Square energy */
+
+/* This subroutine has no local state. */
+
+/* Subroutine */ int energy_(integer *len, real *speech, real *rms)
+{
+    /* System generated locals */
+    integer i__1;
+
+    /* Builtin functions */
+    double sqrt(doublereal);
+
+    /* Local variables */
+    integer i__;
+
+/*       Arguments */
+/*       Local variables that need not be saved */
+    /* Parameter adjustments */
+    --speech;
+
+    /* Function Body */
+    *rms = 0.f;
+    i__1 = *len;
+    for (i__ = 1; i__ <= i__1; ++i__) {
+	*rms += speech[i__] * speech[i__];
+    }
+    *rms = sqrt(*rms / *len);
+    return 0;
+} /* energy_ */
+

codecs/lpc10/f2c.h

@@ -0,0 +1,252 @@
+/*
+
+$Log$
+Revision 1.4  2000/01/05 08:20:39  markster
+Version 0.1.5 from FTP
+
+Revision 1.2  2000/01/05 08:20:39  markster
+Some OSS fixes and a few lpc changes to make it actually work
+
+ * Revision 1.2  1996/08/20  20:26:28  jaf
+ * Any typedef defining a type that was used in lpc10_encoder_state or
+ * lpc10_decoder_state struct's was commented out here and added to
+ * lpc10.h.
+ *
+ * Revision 1.1  1996/08/19  22:32:13  jaf
+ * Initial revision
+ *
+
+*/
+
+/*
+ * f2c.h
+ *
+ * SCCS ID:  @(#)f2c.h 1.2 96/05/19
+ */
+
+/* f2c.h  --  Standard Fortran to C header file */
+
+/**  barf  [ba:rf]  2.  "He suggested using FORTRAN, and everybody barfed."
+
+	- From The Shogakukan DICTIONARY OF NEW ENGLISH (Second edition) */
+
+#ifndef F2C_INCLUDE
+#define F2C_INCLUDE
+
+#include "lpc10.h"
+
+/*typedef long int integer;*/
+/*typedef INT32 integer;*/
+/*typedef short int shortint;*/
+/*typedef INT16 shortint;*/
+/*typedef float real;*/
+/* doublereal only used for function arguments to sqrt, exp, etc. */
+typedef double doublereal;
+/* 32 bits seems wasteful, but there really aren't that many logical
+ * variables around, and making them 32 bits could avoid word
+ * alignment problems, perhaps.  */
+/*typedef long int logical;*/
+/*typedef INT32 logical;*/
+/* The following types are not used in the translated C code for the
+ * LPC-10 coder, but they might be needed by the definitions down
+ * below, so they don't cause compilation errors.  */
+typedef char *address;
+typedef struct { real r, i; } complex;
+typedef struct { doublereal r, i; } doublecomplex;
+typedef short int shortlogical;
+typedef char logical1;
+typedef char integer1;
+/* typedef long long longint; */ /* system-dependent */
+
+#define TRUE_ (1)
+#define FALSE_ (0)
+
+/* Extern is for use with -E */
+#ifndef Extern
+#define Extern extern
+#endif
+
+/* I/O stuff */
+
+#ifdef f2c_i2
+/* for -i2 */
+typedef short flag;
+typedef short ftnlen;
+typedef short ftnint;
+#else
+typedef long int flag;
+typedef long int ftnlen;
+typedef long int ftnint;
+#endif
+
+/*external read, write*/
+typedef struct
+{	flag cierr;
+	ftnint ciunit;
+	flag ciend;
+	char *cifmt;
+	ftnint cirec;
+} cilist;
+
+/*internal read, write*/
+typedef struct
+{	flag icierr;
+	char *iciunit;
+	flag iciend;
+	char *icifmt;
+	ftnint icirlen;
+	ftnint icirnum;
+} icilist;
+
+/*open*/
+typedef struct
+{	flag oerr;
+	ftnint ounit;
+	char *ofnm;
+	ftnlen ofnmlen;
+	char *osta;
+	char *oacc;
+	char *ofm;
+	ftnint orl;
+	char *oblnk;
+} olist;
+
+/*close*/
+typedef struct
+{	flag cerr;
+	ftnint cunit;
+	char *csta;
+} cllist;
+
+/*rewind, backspace, endfile*/
+typedef struct
+{	flag aerr;
+	ftnint aunit;
+} alist;
+
+/* inquire */
+typedef struct
+{	flag inerr;
+	ftnint inunit;
+	char *infile;
+	ftnlen infilen;
+	ftnint	*inex;	/*parameters in standard's order*/
+	ftnint	*inopen;
+	ftnint	*innum;
+	ftnint	*innamed;
+	char	*inname;
+	ftnlen	innamlen;
+	char	*inacc;
+	ftnlen	inacclen;
+	char	*inseq;
+	ftnlen	inseqlen;
+	char 	*indir;
+	ftnlen	indirlen;
+	char	*infmt;
+	ftnlen	infmtlen;
+	char	*inform;
+	ftnint	informlen;
+	char	*inunf;
+	ftnlen	inunflen;
+	ftnint	*inrecl;
+	ftnint	*innrec;
+	char	*inblank;
+	ftnlen	inblanklen;
+} inlist;
+
+#define VOID void
+
+union Multitype {	/* for multiple entry points */
+	integer1 g;
+	shortint h;
+	integer i;
+	/* longint j; */
+	real r;
+	doublereal d;
+	complex c;
+	doublecomplex z;
+	};
+
+typedef union Multitype Multitype;
+
+/*typedef long int Long;*/	/* No longer used; formerly in Namelist */
+
+struct Vardesc {	/* for Namelist */
+	char *name;
+	char *addr;
+	ftnlen *dims;
+	int  type;
+	};
+typedef struct Vardesc Vardesc;
+
+struct Namelist {
+	char *name;
+	Vardesc **vars;
+	int nvars;
+	};
+typedef struct Namelist Namelist;
+
+#define abs(x) ((x) >= 0 ? (x) : -(x))
+#define dabs(x) (doublereal)abs(x)
+#define min(a,b) ((a) <= (b) ? (a) : (b))
+#define max(a,b) ((a) >= (b) ? (a) : (b))
+#define dmin(a,b) (doublereal)min(a,b)
+#define dmax(a,b) (doublereal)max(a,b)
+
+/* procedure parameter types for -A and -C++ */
+
+#define F2C_proc_par_types 1
+#ifdef __cplusplus
+typedef int /* Unknown procedure type */ (*U_fp)(...);
+typedef shortint (*J_fp)(...);
+typedef integer (*I_fp)(...);
+typedef real (*R_fp)(...);
+typedef doublereal (*D_fp)(...), (*E_fp)(...);
+typedef /* Complex */ VOID (*C_fp)(...);
+typedef /* Double Complex */ VOID (*Z_fp)(...);
+typedef logical (*L_fp)(...);
+typedef shortlogical (*K_fp)(...);
+typedef /* Character */ VOID (*H_fp)(...);
+typedef /* Subroutine */ int (*S_fp)(...);
+#else
+typedef int /* Unknown procedure type */ (*U_fp)();
+typedef shortint (*J_fp)();
+typedef integer (*I_fp)();
+typedef real (*R_fp)();
+typedef doublereal (*D_fp)(), (*E_fp)();
+typedef /* Complex */ VOID (*C_fp)();
+typedef /* Double Complex */ VOID (*Z_fp)();
+typedef logical (*L_fp)();
+typedef shortlogical (*K_fp)();
+typedef /* Character */ VOID (*H_fp)();
+typedef /* Subroutine */ int (*S_fp)();
+#endif
+/* E_fp is for real functions when -R is not specified */
+typedef VOID C_f;	/* complex function */
+typedef VOID H_f;	/* character function */
+typedef VOID Z_f;	/* double complex function */
+typedef doublereal E_f;	/* real function with -R not specified */
+
+/* undef any lower-case symbols that your C compiler predefines, e.g.: */
+
+#ifndef Skip_f2c_Undefs
+#undef cray
+#undef gcos
+#undef mc68010
+#undef mc68020
+#undef mips
+#undef pdp11
+#undef sgi
+#undef sparc
+#undef sun
+#undef sun2
+#undef sun3
+#undef sun4
+#undef u370
+#undef u3b
+#undef u3b2
+#undef u3b5
+#undef unix
+#undef vax
+#endif
+#endif

codecs/lpc10/f2clib.c

@@ -0,0 +1,82 @@
+/*
+
+$Log$
+Revision 1.4  2000/01/05 08:20:39  markster
+Version 0.1.5 from FTP
+
+Revision 1.2  2000/01/05 08:20:39  markster
+Some OSS fixes and a few lpc changes to make it actually work
+
+ * Revision 1.1  1996/08/19  22:32:10  jaf
+ * Initial revision
+ *
+
+*/
+
+/*
+ * f2clib.c
+ *
+ * SCCS ID:  @(#)f2clib.c 1.2 96/05/19
+ */
+
+#include "f2c.h"
+
+#ifdef KR_headers
+integer pow_ii(ap, bp) integer *ap, *bp;
+#else
+integer pow_ii(integer *ap, integer *bp)
+#endif
+{
+	integer pow, x, n;
+	unsigned long u;
+
+	x = *ap;
+	n = *bp;
+
+	if (n <= 0) {
+		if (n == 0 || x == 1)
+			return 1;
+		if (x != -1)
+			return x == 0 ? 1/x : 0;
+		n = -n;
+		}
+	u = n;
+	for(pow = 1; ; )
+		{
+		if(u & 01)
+			pow *= x;
+		if(u >>= 1)
+			x *= x;
+		else
+			break;
+		}
+	return(pow);
+	}
+
+
+
+#ifdef KR_headers
+double r_sign(a,b) real *a, *b;
+#else
+double r_sign(real *a, real *b)
+#endif
+{
+double x;
+x = (*a >= 0 ? *a : - *a);
+return( *b >= 0 ? x : -x);
+}
+
+
+
+#ifdef KR_headers
+double floor();
+integer i_nint(x) real *x;
+#else
+#undef abs
+#include "math.h"
+integer i_nint(real *x)
+#endif
+{
+return( (*x)>=0 ?
+	floor(*x + .5) : -floor(.5 - *x) );
+}

codecs/lpc10/ham84.c

@@ -0,0 +1,114 @@
+/*
+
+$Log$
+Revision 1.4  2000/01/05 08:20:39  markster
+Version 0.1.5 from FTP
+
+Revision 1.2  2000/01/05 08:20:39  markster
+Some OSS fixes and a few lpc changes to make it actually work
+
+ * Revision 1.1  1996/08/19  22:32:07  jaf
+ * Initial revision
+ *
+
+*/
+
+#ifdef P_R_O_T_O_T_Y_P_E_S
+extern int ham84_(integer *input, integer *output, integer *errcnt);
+#endif
+
+/*  -- translated by f2c (version 19951025).
+   You must link the resulting object file with the libraries:
+	-lf2c -lm   (in that order)
+*/
+
+#include "f2c.h"
+
+/* ***************************************************************** */
+
+/* 	HAM84 Version 45G */
+
+/* $Log$
+ * Revision 1.4  2000/01/05 08:20:39  markster
+ * Version 0.1.5 from FTP
+ *
+/* Revision 1.2  2000/01/05 08:20:39  markster
+/* Some OSS fixes and a few lpc changes to make it actually work
+/*
+ * Revision 1.1  1996/08/19  22:32:07  jaf
+ * Initial revision
+ * */
+/* Revision 1.3  1996/03/21  15:26:00  jaf */
+/* Put comment header in standard form. */
+
+/* Revision 1.2  1996/03/13  22:00:13  jaf */
+/* Comments added explaining that none of the local variables of this */
+/* subroutine need to be saved from one invocation to the next. */
+
+/* Revision 1.1  1996/02/07 14:47:04  jaf */
+/* Initial revision */
+
+
+/* ***************************************************************** */
+
+/*  Hamming 8,4 Decoder - can correct 1 out of seven bits */
+/*   and can detect up to two errors. */
+
+/* Input: */
+/*  INPUT  - Seven bit data word, 4 bits parameter and */
+/*           4 bits parity information */
+/* Input/Output: */
+/*  ERRCNT - Sums errors detected by Hamming code */
+/* Output: */
+/*  OUTPUT - 4 corrected parameter bits */
+
+/* This subroutine is entered with an eight bit word in INPUT.  The 8th */
+/* bit is parity and is stripped off.  The remaining 7 bits address the */
+/* hamming 8,4 table and the output OUTPUT from the table gives the 4 */
+/* bits of corrected data.  If bit 4 is set, no error was detected. */
+/* ERRCNT is the number of errors counted. */
+
+/* This subroutine has no local state. */
+
+/* Subroutine */ int ham84_(integer *input, integer *output, integer *errcnt)
+{
+    /* Initialized data */
+
+    static integer dactab[128] = { 16,0,0,3,0,5,14,7,0,9,14,11,14,13,30,14,0,
+	    9,2,7,4,7,7,23,9,25,10,9,12,9,14,7,0,5,2,11,5,21,6,5,8,11,11,27,
+	    12,5,14,11,2,1,18,2,12,5,2,7,12,9,2,11,28,12,12,15,0,3,3,19,4,13,
+	    6,3,8,13,10,3,13,29,14,13,4,1,10,3,20,4,4,7,10,9,26,10,4,13,10,15,
+	    8,1,6,3,6,5,22,6,24,8,8,11,8,13,6,15,1,17,2,1,4,1,6,15,8,1,10,15,
+	    12,15,15,31 };
+
+    integer i__, j, parity;
+
+/*       Arguments */
+/*       Parameters/constants */
+/*       Local variables that need not be saved */
+/*  Determine parity of input word */
+    parity = *input & 255;
+    parity ^= parity / 16;
+    parity ^= parity / 4;
+    parity ^= parity / 2;
+    parity &= 1;
+    i__ = dactab[*input & 127];
+    *output = i__ & 15;
+    j = i__ & 16;
+    if (j != 0) {
+/*          No errors detected in seven bits */
+	if (parity != 0) {
+	    ++(*errcnt);
+	}
+    } else {
+/*          One or two errors detected */
+	++(*errcnt);
+	if (parity == 0) {
+/*             Two errors detected */
+	    ++(*errcnt);
+	    *output = -1;
+	}
+    }
+    return 0;
+} /* ham84_ */
+

codecs/lpc10/hp100.c

@@ -0,0 +1,157 @@
+/*
+
+$Log$
+Revision 1.4  2000/01/05 08:20:39  markster
+Version 0.1.5 from FTP
+
+Revision 1.2  2000/01/05 08:20:39  markster
+Some OSS fixes and a few lpc changes to make it actually work
+
+ * Revision 1.2  1996/08/20  20:28:05  jaf
+ * Removed all static local variables that were SAVE'd in the Fortran
+ * code, and put them in struct lpc10_encoder_state that is passed as an
+ * argument.
+ *
+ * Removed init function, since all initialization is now done in
+ * init_lpc10_encoder_state().
+ *
+ * Revision 1.1  1996/08/19  22:32:04  jaf
+ * Initial revision
+ *
+
+*/
+
+#ifdef P_R_O_T_O_T_Y_P_E_S
+extern int hp100_(real *speech, integer *start, integer *end,
+		  struct lpc10_encoder_state *st);
+extern int inithp100_(void);
+#endif
+
+/*  -- translated by f2c (version 19951025).
+   You must link the resulting object file with the libraries:
+	-lf2c -lm   (in that order)
+*/
+
+#include "f2c.h"
+
+/* ********************************************************************* */
+
+/*      HP100 Version 55 */
+
+/* $Log$
+ * Revision 1.4  2000/01/05 08:20:39  markster
+ * Version 0.1.5 from FTP
+ *
+/* Revision 1.2  2000/01/05 08:20:39  markster
+/* Some OSS fixes and a few lpc changes to make it actually work
+/*
+ * Revision 1.2  1996/08/20  20:28:05  jaf
+ * Removed all static local variables that were SAVE'd in the Fortran
+ * code, and put them in struct lpc10_encoder_state that is passed as an
+ * argument.
+ *
+ * Removed init function, since all initialization is now done in
+ * init_lpc10_encoder_state().
+ *
+ * Revision 1.1  1996/08/19  22:32:04  jaf
+ * Initial revision
+ * */
+/* Revision 1.6  1996/03/15  16:45:25  jaf */
+/* Rearranged a few comments. */
+
+/* Revision 1.5  1996/03/14  23:20:54  jaf */
+/* Added comments about when INITHP100 should be used. */
+
+/* Revision 1.4  1996/03/14  23:08:08  jaf */
+/* Added an entry named INITHP100 that initializes the local state of */
+/* subroutine HP100. */
+
+/* Revision 1.3  1996/03/14  22:09:20  jaf */
+/* Comments added explaining which of the local variables of this */
+/* subroutine need to be saved from one invocation to the next, and which */
+/* do not. */
+
+/* Revision 1.2  1996/02/12  15:05:54  jaf */
+/* Added lots of comments explaining why I changed one line, which was a */
+/* declaration with initializations. */
+
+/* Revision 1.1  1996/02/07 14:47:12  jaf */
+/* Initial revision */
+
+
+/* ********************************************************************* */
+
+/*    100 Hz High Pass Filter */
+
+/* Jan 92 - corrected typo (1.937148 to 1.935715), */
+/*          rounded coefficients to 7 places, */
+/*          corrected and merged gain (.97466**4), */
+/*          merged numerator into first two sections. */
+
+/* Input: */
+/*  start, end - Range of samples to filter */
+/* Input/Output: */
+/*  speech(end) - Speech data. */
+/*                Indices start through end are read and modified. */
+
+/* This subroutine maintains local state from one call to the next.  If */
+/* you want to switch to using a new audio stream for this filter, or */
+/* reinitialize its state for any other reason, call the ENTRY */
+/* INITHP100. */
+/* Subroutine */ int hp100_(real *speech, integer *start, integer *end,
+	struct lpc10_encoder_state *st)
+{
+    /* Temporary local copies of variables in lpc10_encoder_state.
+       I've only created these because it might cause the loop below
+       to execute a bit faster to access local variables, rather than
+       variables in the lpc10_encoder_state structure.  It is just a
+       guess that it will be faster. */
+
+    real z11;
+    real z21;
+    real z12;
+    real z22;
+
+    /* System generated locals */
+    integer i__1;
+
+    /* Local variables */
+    integer i__;
+    real si, err;
+
+/*       Arguments */
+/*       Local variables that need not be saved */
+/*       Local state */
+    /* Parameter adjustments */
+    if (speech) {
+	--speech;
+	}
+
+    /* Function Body */
+
+    z11 = st->z11;
+    z21 = st->z21;
+    z12 = st->z12;
+    z22 = st->z22;
+
+    i__1 = *end;
+    for (i__ = *start; i__ <= i__1; ++i__) {
+	si = speech[i__];
+	err = si + z11 * 1.859076f - z21 * .8648249f;
+	si = err - z11 * 2.f + z21;
+	z21 = z11;
+	z11 = err;
+	err = si + z12 * 1.935715f - z22 * .9417004f;
+	si = err - z12 * 2.f + z22;
+	z22 = z12;
+	z12 = err;
+	speech[i__] = si * .902428f;
+    }
+
+    st->z11 = z11;
+    st->z21 = z21;
+    st->z12 = z12;
+    st->z22 = z22;
+
+    return 0;
+} /* hp100_ */

codecs/lpc10/invert.c

@@ -0,0 +1,175 @@
+/*
+
+$Log$
+Revision 1.4  2000/01/05 08:20:39  markster
+Version 0.1.5 from FTP
+
+Revision 1.2  2000/01/05 08:20:39  markster
+Some OSS fixes and a few lpc changes to make it actually work
+
+ * Revision 1.1  1996/08/19  22:32:00  jaf
+ * Initial revision
+ *
+
+*/
+
+#ifdef P_R_O_T_O_T_Y_P_E_S
+extern int invert_(integer *order, real *phi, real *psi, real *rc);
+#endif
+
+/*  -- translated by f2c (version 19951025).
+   You must link the resulting object file with the libraries:
+	-lf2c -lm   (in that order)
+*/
+
+#include "f2c.h"
+
+/* **************************************************************** */
+
+/* 	INVERT Version 45G */
+
+/* $Log$
+ * Revision 1.4  2000/01/05 08:20:39  markster
+ * Version 0.1.5 from FTP
+ *
+/* Revision 1.2  2000/01/05 08:20:39  markster
+/* Some OSS fixes and a few lpc changes to make it actually work
+/*
+ * Revision 1.1  1996/08/19  22:32:00  jaf
+ * Initial revision
+ * */
+/* Revision 1.3  1996/03/18  20:52:47  jaf */
+/* Just added a few comments about which array indices of the arguments */
+/* are used, and mentioning that this subroutine has no local state. */
+
+/* Revision 1.2  1996/03/13  16:51:32  jaf */
+/* Comments added explaining that none of the local variables of this */
+/* subroutine need to be saved from one invocation to the next. */
+
+/* Eliminated a comment from the original, describing a local array X */
+/* that appeared nowhere in the code. */
+
+/* Revision 1.1  1996/02/07 14:47:20  jaf */
+/* Initial revision */
+
+
+/* **************************************************************** */
+
+/*  Invert a covariance matrix using Choleski decomposition method. */
+
+/* Input: */
+/*  ORDER            - Analysis order */
+/*  PHI(ORDER,ORDER) - Covariance matrix */
+/*                    Indices (I,J) read, where ORDER .GE. I .GE. J .GE. 1.*/
+/*                     All other indices untouched. */
+/*  PSI(ORDER)       - Column vector to be predicted */
+/*                     Indices 1 through ORDER read. */
+/* Output: */
+/*  RC(ORDER)        - Pseudo reflection coefficients */
+/*                    Indices 1 through ORDER written, and then possibly read.
+*/
+/* Internal: */
+/*  V(ORDER,ORDER)   - Temporary matrix */
+/*                     Same indices written as read from PHI. */
+/*                     Many indices may be read and written again after */
+/*                     initially being copied from PHI, but all indices */
+/*                     are written before being read. */
+
+/*  NOTE: Temporary matrix V is not needed and may be replaced */
+/*    by PHI if the original PHI values do not need to be preserved. */
+
+/* Subroutine */ int invert_(integer *order, real *phi, real *psi, real *rc)
+{
+    /* System generated locals */
+    integer phi_dim1, phi_offset, i__1, i__2, i__3;
+    real r__1, r__2;
+
+    /* Local variables */
+    real save;
+    integer i__, j, k;
+    real v[100]	/* was [10][10] */;
+
+/*       Arguments */
+/* $Log$
+ * Revision 1.4  2000/01/05 08:20:39  markster
+ * Version 0.1.5 from FTP
+ *
+/* Revision 1.2  2000/01/05 08:20:39  markster
+/* Some OSS fixes and a few lpc changes to make it actually work
+/*
+ * Revision 1.1  1996/08/19  22:32:00  jaf
+ * Initial revision
+ * */
+/* Revision 1.3  1996/03/29  22:03:47  jaf */
+/* Removed definitions for any constants that were no longer used. */
+
+/* Revision 1.2  1996/03/26  19:34:33  jaf */
+/* Added comments indicating which constants are not needed in an */
+/* application that uses the LPC-10 coder. */
+
+/* Revision 1.1  1996/02/07  14:43:51  jaf */
+/* Initial revision */
+
+/*   LPC Configuration parameters: */
+/* Frame size, Prediction order, Pitch period */
+/* 	Parameters/constants */
+/*       Local variables that need not be saved */
+/*  Decompose PHI into V * D * V' where V is a triangular matrix whose */
+/*   main diagonal elements are all 1, V' is the transpose of V, and */
+/*   D is a vector.  Here D(n) is stored in location V(n,n). */
+    /* Parameter adjustments */
+    --rc;
+    --psi;
+    phi_dim1 = *order;
+    phi_offset = phi_dim1 + 1;
+    phi -= phi_offset;
+
+    /* Function Body */
+    i__1 = *order;
+    for (j = 1; j <= i__1; ++j) {
+	i__2 = *order;
+	for (i__ = j; i__ <= i__2; ++i__) {
+	    v[i__ + j * 10 - 11] = phi[i__ + j * phi_dim1];
+	}
+	i__2 = j - 1;
+	for (k = 1; k <= i__2; ++k) {
+	    save = v[j + k * 10 - 11] * v[k + k * 10 - 11];
+	    i__3 = *order;
+	    for (i__ = j; i__ <= i__3; ++i__) {
+		v[i__ + j * 10 - 11] -= v[i__ + k * 10 - 11] * save;
+	    }
+	}
+/*  Compute intermediate results, which are similar to RC's */
+	if ((r__1 = v[j + j * 10 - 11], abs(r__1)) < 1e-10f) {
+	    goto L100;
+	}
+	rc[j] = psi[j];
+	i__2 = j - 1;
+	for (k = 1; k <= i__2; ++k) {
+	    rc[j] -= rc[k] * v[j + k * 10 - 11];
+	}
+	v[j + j * 10 - 11] = 1.f / v[j + j * 10 - 11];
+	rc[j] *= v[j + j * 10 - 11];
+/* Computing MAX */
+/* Computing MIN */
+	r__2 = rc[j];
+	r__1 = min(r__2,.999f);
+	rc[j] = max(r__1,-.999f);
+    }
+    return 0;
+/*  Zero out higher order RC's if algorithm terminated early */
+L100:
+    i__1 = *order;
+    for (i__ = j; i__ <= i__1; ++i__) {
+	rc[i__] = 0.f;
+    }
+/*  Back substitute for PC's (if needed) */
+/* 110	DO J = ORDER,1,-1 */
+/* 	   PC(J) = RC(J) */
+/* 	   DO I = 1,J-1 */
+/* 	      PC(J) = PC(J) - PC(I)*V(J,I) */
+/* 	   END DO */
+/* 	END DO */
+    return 0;
+} /* invert_ */
+

codecs/lpc10/irc2pc.c

@@ -0,0 +1,133 @@
+/*
+
+$Log$
+Revision 1.4  2000/01/05 08:20:39  markster
+Version 0.1.5 from FTP
+
+Revision 1.2  2000/01/05 08:20:39  markster
+Some OSS fixes and a few lpc changes to make it actually work
+
+ * Revision 1.1  1996/08/19  22:31:56  jaf
+ * Initial revision
+ *
+
+*/
+
+#ifdef P_R_O_T_O_T_Y_P_E_S
+extern int irc2pc_(real *rc, real *pc, integer *order, real *gprime, real *g2pass);
+#endif
+
+/*  -- translated by f2c (version 19951025).
+   You must link the resulting object file with the libraries:
+	-lf2c -lm   (in that order)
+*/
+
+#include "f2c.h"
+
+/* ***************************************************************** */
+
+/* 	IRC2PC Version 48 */
+
+/* $Log$
+ * Revision 1.4  2000/01/05 08:20:39  markster
+ * Version 0.1.5 from FTP
+ *
+/* Revision 1.2  2000/01/05 08:20:39  markster
+/* Some OSS fixes and a few lpc changes to make it actually work
+/*
+ * Revision 1.1  1996/08/19  22:31:56  jaf
+ * Initial revision
+ * */
+/* Revision 1.3  1996/03/20  15:47:19  jaf */
+/* Added comments about which indices of array arguments are read or */
+/* written. */
+
+/* Revision 1.2  1996/03/14  16:59:04  jaf */
+/* Comments added explaining that none of the local variables of this */
+/* subroutine need to be saved from one invocation to the next. */
+
+/* Revision 1.1  1996/02/07 14:47:27  jaf */
+/* Initial revision */
+
+
+/* ***************************************************************** */
+
+/*   Convert Reflection Coefficients to Predictor Coeficients */
+
+/* Inputs: */
+/*  RC     - Reflection coefficients */
+/*           Indices 1 through ORDER read. */
+/*  ORDER  - Number of RC's */
+/*  GPRIME - Excitation modification gain */
+/* Outputs: */
+/*  PC     - Predictor coefficients */
+/*           Indices 1 through ORDER written. */
+/*           Indices 1 through ORDER-1 are read after being written. */
+/*  G2PASS - Excitation modification sharpening factor */
+
+/* This subroutine has no local state. */
+
+/* Subroutine */ int irc2pc_(real *rc, real *pc, integer *order, real *gprime,
+	 real *g2pass)
+{
+    /* System generated locals */
+    integer i__1, i__2;
+
+    /* Builtin functions */
+    double sqrt(doublereal);
+
+    /* Local variables */
+    real temp[10];
+    integer i__, j;
+
+/* 	Arguments */
+/* $Log$
+ * Revision 1.4  2000/01/05 08:20:39  markster
+ * Version 0.1.5 from FTP
+ *
+/* Revision 1.2  2000/01/05 08:20:39  markster
+/* Some OSS fixes and a few lpc changes to make it actually work
+/*
+ * Revision 1.1  1996/08/19  22:31:56  jaf
+ * Initial revision
+ * */
+/* Revision 1.3  1996/03/29  22:03:47  jaf */
+/* Removed definitions for any constants that were no longer used. */
+
+/* Revision 1.2  1996/03/26  19:34:33  jaf */
+/* Added comments indicating which constants are not needed in an */
+/* application that uses the LPC-10 coder. */
+
+/* Revision 1.1  1996/02/07  14:43:51  jaf */
+/* Initial revision */
+
+/*   LPC Configuration parameters: */
+/* Frame size, Prediction order, Pitch period */
+/*       Local variables that need not be saved */
+    /* Parameter adjustments */
+    --pc;
+    --rc;
+
+    /* Function Body */
+    *g2pass = 1.f;
+    i__1 = *order;
+    for (i__ = 1; i__ <= i__1; ++i__) {
+	*g2pass *= 1.f - rc[i__] * rc[i__];
+    }
+    *g2pass = *gprime * sqrt(*g2pass);
+    pc[1] = rc[1];
+    i__1 = *order;
+    for (i__ = 2; i__ <= i__1; ++i__) {
+	i__2 = i__ - 1;
+	for (j = 1; j <= i__2; ++j) {
+	    temp[j - 1] = pc[j] - rc[i__] * pc[i__ - j];
+	}
+	i__2 = i__ - 1;
+	for (j = 1; j <= i__2; ++j) {
+	    pc[j] = temp[j - 1];
+	}
+	pc[i__] = rc[i__];
+    }
+    return 0;
+} /* irc2pc_ */
+

codecs/lpc10/ivfilt.c

@@ -0,0 +1,118 @@
+/*
+
+$Log$
+Revision 1.4  2000/01/05 08:20:39  markster
+Version 0.1.5 from FTP
+
+Revision 1.2  2000/01/05 08:20:39  markster
+Some OSS fixes and a few lpc changes to make it actually work
+
+ * Revision 1.1  1996/08/19  22:31:53  jaf
+ * Initial revision
+ *
+
+*/
+
+#ifdef P_R_O_T_O_T_Y_P_E_S
+extern int ivfilt_(real *lpbuf, real *ivbuf, integer *len, integer *nsamp, real *ivrc);
+#endif
+
+/*  -- translated by f2c (version 19951025).
+   You must link the resulting object file with the libraries:
+	-lf2c -lm   (in that order)
+*/
+
+#include "f2c.h"
+
+/* ********************************************************************* */
+
+/* 	IVFILT Version 48 */
+
+/* $Log$
+ * Revision 1.4  2000/01/05 08:20:39  markster
+ * Version 0.1.5 from FTP
+ *
+/* Revision 1.2  2000/01/05 08:20:39  markster
+/* Some OSS fixes and a few lpc changes to make it actually work
+/*
+ * Revision 1.1  1996/08/19  22:31:53  jaf
+ * Initial revision
+ * */
+/* Revision 1.3  1996/03/15  21:36:29  jaf */
+/* Just added a few comments about which array indices of the arguments */
+/* are used, and mentioning that this subroutine has no local state. */
+
+/* Revision 1.2  1996/03/13  00:01:00  jaf */
+/* Comments added explaining that none of the local variables of this */
+/* subroutine need to be saved from one invocation to the next. */
+
+/* Revision 1.1  1996/02/07 14:47:34  jaf */
+/* Initial revision */
+
+
+/* ********************************************************************* */
+
+/*   2nd order inverse filter, speech is decimated 4:1 */
+
+/* Input: */
+/*  LEN    - Length of speech buffers */
+/*  NSAMP  - Number of samples to filter */
+/*  LPBUF  - Low pass filtered speech buffer */
+/*           Indices LEN-NSAMP-7 through LEN read. */
+/* Output: */
+/*  IVBUF  - Inverse filtered speech buffer */
+/*           Indices LEN-NSAMP+1 through LEN written. */
+/*  IVRC   - Inverse filter reflection coefficients (for voicing) */
+/*          Indices 1 and 2 both written (also read, but only after writing).
+*/
+
+/* This subroutine has no local state. */
+
+/* Subroutine */ int ivfilt_(real *lpbuf, real *ivbuf, integer *len, integer *
+	nsamp, real *ivrc)
+{
+    /* System generated locals */
+    integer i__1;
+
+    /* Local variables */
+    integer i__, j, k;
+    real r__[3], pc1, pc2;
+
+/* 	Arguments */
+/*       Local variables that need not be saved */
+/*       Local state */
+/*       None */
+/*  Calculate Autocorrelations */
+    /* Parameter adjustments */
+    --ivbuf;
+    --lpbuf;
+    --ivrc;
+
+    /* Function Body */
+    for (i__ = 1; i__ <= 3; ++i__) {
+	r__[i__ - 1] = 0.f;
+	k = i__ - 1 << 2;
+	i__1 = *len;
+	for (j = (i__ << 2) + *len - *nsamp; j <= i__1; j += 2) {
+	    r__[i__ - 1] += lpbuf[j] * lpbuf[j - k];
+	}
+    }
+/*  Calculate predictor coefficients */
+    pc1 = 0.f;
+    pc2 = 0.f;
+    ivrc[1] = 0.f;
+    ivrc[2] = 0.f;
+    if (r__[0] > 1e-10f) {
+	ivrc[1] = r__[1] / r__[0];
+	ivrc[2] = (r__[2] - ivrc[1] * r__[1]) / (r__[0] - ivrc[1] * r__[1]);
+	pc1 = ivrc[1] - ivrc[1] * ivrc[2];
+	pc2 = ivrc[2];
+    }
+/*  Inverse filter LPBUF into IVBUF */
+    i__1 = *len;
+    for (i__ = *len + 1 - *nsamp; i__ <= i__1; ++i__) {
+	ivbuf[i__] = lpbuf[i__] - pc1 * lpbuf[i__ - 4] - pc2 * lpbuf[i__ - 8];
+    }
+    return 0;
+} /* ivfilt_ */
+

codecs/lpc10/lpc10.h

@@ -0,0 +1,229 @@
+/*
+
+$Log$
+Revision 1.4  2000/01/05 00:20:06  markster
+Version 0.1.5 from FTP
+
+Revision 1.1  2000/01/05 00:20:06  markster
+Add broken lpc10 code...  It's not too far from working I don't think...
+
+ * Revision 1.1  1996/08/19  22:47:31  jaf
+ * Initial revision
+ *
+
+*/
+
+#ifndef __LPC10_H__
+#define __LPC10_H__
+
+#define LPC10_SAMPLES_PER_FRAME 180
+#define LPC10_BITS_IN_COMPRESSED_FRAME 54
+
+
+/*
+
+  The "#if defined"'s in this file are by no means intended to be
+  complete.  They are what Nautilus uses, which has been successfully
+  compiled under DOS with the Microsoft C compiler, and under a few
+  versions of Unix with the GNU C compiler.
+
+ */
+
+#if defined(unix)
+typedef short		INT16;
+typedef int		INT32;
+#endif
+
+
+#if defined(__MSDOS__) || defined(MSDOS)
+typedef int		INT16;
+typedef long		INT32;
+#endif
+
+
+
+/* The initial values for every member of this structure is 0, except
+   where noted in comments. */
+
+/* These two lines are copied from f2c.h.  There should be a more
+   elegant way of doing this than having the same declarations in two
+   files. */
+
+typedef float real;
+typedef INT32 integer;
+typedef INT32 logical;
+typedef INT16 shortint;
+
+struct lpc10_encoder_state {
+    /* State used only by function hp100 */
+    real z11;
+    real z21;
+    real z12;
+    real z22;
+    
+    /* State used by function analys */
+    real inbuf[540], pebuf[540];
+    real lpbuf[696], ivbuf[312];
+    real bias;
+    integer osbuf[10];  /* no initial value necessary */
+    integer osptr;     /* initial value 1 */
+    integer obound[3];
+    integer vwin[6]	/* was [2][3] */;   /* initial value vwin[4] = 307; vwin[5] = 462; */
+    integer awin[6]	/* was [2][3] */;   /* initial value awin[4] = 307; awin[5] = 462; */
+    integer voibuf[8]	/* was [2][4] */;
+    real rmsbuf[3];
+    real rcbuf[30]	/* was [10][3] */;
+    real zpre;
+
+
+    /* State used by function onset */
+    real n;
+    real d__;   /* initial value 1.f */
+    real fpc;   /* no initial value necessary */
+    real l2buf[16];
+    real l2sum1;
+    integer l2ptr1;   /* initial value 1 */
+    integer l2ptr2;   /* initial value 9 */
+    integer lasti;    /* no initial value necessary */
+    logical hyst;   /* initial value FALSE_ */
+
+    /* State used by function voicin */
+    real dither;   /* initial value 20.f */
+    real snr;
+    real maxmin;
+    real voice[6]	/* was [2][3] */;   /* initial value is probably unnecessary */
+    integer lbve, lbue, fbve, fbue;
+    integer ofbue, sfbue;
+    integer olbue, slbue;
+    /* Initial values:
+	lbve = 3000;
+	fbve = 3000;
+	fbue = 187;
+	ofbue = 187;
+	sfbue = 187;
+	lbue = 93;
+	olbue = 93;
+	slbue = 93;
+	snr = (real) (fbve / fbue << 6);
+	*/
+
+    /* State used by function dyptrk */
+    real s[60];
+    integer p[120]	/* was [60][2] */;
+    integer ipoint;
+    real alphax;
+
+    /* State used by function chanwr */
+    integer isync;
+
+};
+
+
+struct lpc10_decoder_state {
+
+    /* State used by function decode */
+    integer iptold;   /* initial value 60 */
+    logical first;   /* initial value TRUE_ */
+    integer ivp2h;
+    integer iovoic;
+    integer iavgp;   /* initial value 60 */
+    integer erate;
+    integer drc[30]	/* was [3][10] */;
+    integer dpit[3];
+    integer drms[3];
+
+    /* State used by function synths */
+    real buf[360];
+    integer buflen;   /* initial value 180 */
+
+    /* State used by function pitsyn */
+    integer ivoico;   /* no initial value necessary as long as first_pitsyn is initially TRUE_ */
+    integer ipito;   /* no initial value necessary as long as first_pitsyn is initially TRUE_ */
+    real rmso;   /* initial value 1.f */
+    real rco[10];   /* no initial value necessary as long as first_pitsyn is initially TRUE_ */
+    integer jsamp;   /* no initial value necessary as long as first_pitsyn is initially TRUE_ */
+    logical first_pitsyn;   /* initial value TRUE_ */
+
+    /* State used by function bsynz */
+    integer ipo;
+    real exc[166];
+    real exc2[166];
+    real lpi1;
+    real lpi2;
+    real lpi3;
+    real hpi1;
+    real hpi2;
+    real hpi3;
+    real rmso_bsynz;
+
+    /* State used by function random */
+    integer j;   /* initial value 2 */
+    integer k;   /* initial value 5 */
+    shortint y[5];  /* initial value { -21161,-8478,30892,-10216,16950 } */
+
+    /* State used by function deemp */
+    real dei1;
+    real dei2;
+    real deo1;
+    real deo2;
+    real deo3;
+
+};
+
+
+
+/*
+
+  Calling sequence:
+
+  Call create_lpc10_encoder_state(), which returns a pointer to an
+  already initialized lpc10_encoder_state structure.
+
+  lpc10_encode reads indices 0 through (LPC10_SAMPLES_PER_FRAME-1) of
+  array speech[], and writes indices 0 through
+  (LPC10_BITS_IN_COMPRESSED_FRAME-1) of array bits[], and both reads
+  and writes the lpc10_encoder_state structure contents.  The
+  lpc10_encoder_state structure should *not* be initialized for every
+  frame of encoded speech.  Once at the beginning of execution, done
+  automatically for you by create_lpc10_encoder_state(), is enough.
+
+  init_lpc10_encoder_state() reinitializes the lpc10_encoder_state
+  structure.  This might be useful if you are finished processing one
+  sound sample, and want to reuse the same lpc10_encoder_state
+  structure to process another sound sample.  There might be other
+  uses as well.
+
+  Note that the comments in the lpc10/lpcenc.c file imply that indices
+  1 through 180 of array speech[] are read.  These comments were
+  written for the Fortran version of the code, before it was
+  automatically converted to C by the conversion program f2c.  f2c
+  seems to use the convention that the pointers to arrays passed as
+  function arguments point to the first index used in the Fortran
+  code, whatever index that might be (usually 1), and then it modifies
+  the pointer inside of the function, like so:
+
+  if (speech) {
+      --speech;
+  }
+
+  So that the code can access the first value at index 1 and the last
+  at index 180.  This makes the translated C code "closer" to the
+  original Fortran code.
+
+  The calling sequence for the decoder is similar to the encoder.  The
+  only significant difference is that the array bits[] is read
+  (indices 0 through (LPC10_BITS_IN_COMPRESSED_FRAME-1)), and the
+  array speech[] is written (indices 0 through
+  (LPC10_SAMPLES_PER_FRAME-1)).
+  
+  */
+
+struct lpc10_encoder_state * create_lpc10_encoder_state ();
+void init_lpc10_encoder_state (struct lpc10_encoder_state *st);
+int lpc10_encode (real *speech, INT32 *bits, struct lpc10_encoder_state *st);
+
+struct lpc10_decoder_state * create_lpc10_decoder_state ();
+void init_lpc10_decoder_state (struct lpc10_decoder_state *st);
+int lpc10_decode (INT32 *bits, real *speech, struct lpc10_decoder_state *st);
+
+#endif /* __LPC10_H__ */

codecs/lpc10/lpcdec.c

@@ -0,0 +1,273 @@
+/*
+
+$Log$
+Revision 1.4  2000/01/05 08:20:39  markster
+Version 0.1.5 from FTP
+
+Revision 1.2  2000/01/05 08:20:39  markster
+Some OSS fixes and a few lpc changes to make it actually work
+
+ * Revision 1.2  1996/08/20  20:30:11  jaf
+ * Removed all static local variables that were SAVE'd in the Fortran
+ * code, and put them in struct lpc10_encoder_state that is passed as an
+ * argument.
+ *
+ * Removed init function, since all initialization is now done in
+ * init_lpc10_encoder_state().
+ *
+ * Changed name of function from lpcenc_ to lpc10_encode, simply to make
+ * all lpc10 functions have more consistent naming with each other.
+ *
+ * Revision 1.1  1996/08/19  22:31:48  jaf
+ * Initial revision
+ *
+
+*/
+
+#ifdef P_R_O_T_O_T_Y_P_E_S
+extern int lpcdec_(integer *bits, real *speech);
+extern int initlpcdec_(void);
+/* comlen contrl_ 12 */
+/*:ref: chanrd_ 14 5 4 4 4 4 4 */
+/*:ref: decode_ 14 7 4 4 4 4 4 6 6 */
+/*:ref: synths_ 14 6 4 4 6 6 6 4 */
+/*:ref: initdecode_ 14 0 */
+/*:ref: initsynths_ 14 0 */
+#endif
+
+/*  -- translated by f2c (version 19951025).
+   You must link the resulting object file with the libraries:
+	-lf2c -lm   (in that order)
+*/
+
+#include "f2c.h"
+
+/* Common Block Declarations */
+
+extern struct {
+    integer order, lframe;
+    logical corrp;
+} contrl_;
+
+#define contrl_1 contrl_
+
+/* Table of constant values */
+
+static integer c__10 = 10;
+
+/* ***************************************************************** */
+
+/* $Log$
+ * Revision 1.4  2000/01/05 08:20:39  markster
+ * Version 0.1.5 from FTP
+ *
+/* Revision 1.2  2000/01/05 08:20:39  markster
+/* Some OSS fixes and a few lpc changes to make it actually work
+/*
+ * Revision 1.2  1996/08/20  20:30:11  jaf
+ * Removed all static local variables that were SAVE'd in the Fortran
+ * code, and put them in struct lpc10_encoder_state that is passed as an
+ * argument.
+ *
+ * Removed init function, since all initialization is now done in
+ * init_lpc10_encoder_state().
+ *
+ * Changed name of function from lpcenc_ to lpc10_encode, simply to make
+ * all lpc10 functions have more consistent naming with each other.
+ *
+ * Revision 1.1  1996/08/19  22:31:48  jaf
+ * Initial revision
+ * */
+/* Revision 1.1  1996/03/28  00:03:00  jaf */
+/* Initial revision */
+
+
+/* ***************************************************************** */
+
+/* Decode 54 bits to one frame of 180 speech samples. */
+
+/* Input: */
+/*  BITS   - 54 encoded bits, stored 1 per array element. */
+/*           Indices 1 through 53 read (SYNC bit ignored). */
+/* Output: */
+/*  SPEECH - Speech encoded as real values in the range [-1,+1]. */
+/*           Indices 1 through 180 written. */
+
+/* This subroutine maintains local state from one call to the next.  If */
+/* you want to switch to using a new audio stream for this filter, or */
+/* reinitialize its state for any other reason, call the ENTRY */
+/* INITLPCDEC. */
+
+/* Subroutine */ int lpc10_decode(integer *bits, real *speech,
+				  struct lpc10_decoder_state *st)
+{
+    integer irms, voice[2], pitch, ipitv;
+    extern /* Subroutine */ int decode_(integer *, integer *, integer *, 
+	    integer *, integer *, real *, real *, struct lpc10_decoder_state *);
+    real rc[10];
+    extern /* Subroutine */ int chanrd_(integer *, integer *, integer *, 
+	    integer *, integer *), synths_(integer *, 
+	    integer *, real *, real *, real *, integer *,
+					   struct lpc10_decoder_state *);
+    integer irc[10], len;
+    real rms;
+
+/* $Log$
+ * Revision 1.4  2000/01/05 08:20:39  markster
+ * Version 0.1.5 from FTP
+ *
+/* Revision 1.2  2000/01/05 08:20:39  markster
+/* Some OSS fixes and a few lpc changes to make it actually work
+/*
+ * Revision 1.2  1996/08/20  20:30:11  jaf
+ * Removed all static local variables that were SAVE'd in the Fortran
+ * code, and put them in struct lpc10_encoder_state that is passed as an
+ * argument.
+ *
+ * Removed init function, since all initialization is now done in
+ * init_lpc10_encoder_state().
+ *
+ * Changed name of function from lpcenc_ to lpc10_encode, simply to make
+ * all lpc10 functions have more consistent naming with each other.
+ *
+ * Revision 1.1  1996/08/19  22:31:48  jaf
+ * Initial revision
+ * */
+/* Revision 1.3  1996/03/29  22:03:47  jaf */
+/* Removed definitions for any constants that were no longer used. */
+
+/* Revision 1.2  1996/03/26  19:34:33  jaf */
+/* Added comments indicating which constants are not needed in an */
+/* application that uses the LPC-10 coder. */
+
+/* Revision 1.1  1996/02/07  14:43:51  jaf */
+/* Initial revision */
+
+/*   LPC Configuration parameters: */
+/* Frame size, Prediction order, Pitch period */
+/*       Arguments */
+/* $Log$
+ * Revision 1.4  2000/01/05 08:20:39  markster
+ * Version 0.1.5 from FTP
+ *
+/* Revision 1.2  2000/01/05 08:20:39  markster
+/* Some OSS fixes and a few lpc changes to make it actually work
+/*
+ * Revision 1.2  1996/08/20  20:30:11  jaf
+ * Removed all static local variables that were SAVE'd in the Fortran
+ * code, and put them in struct lpc10_encoder_state that is passed as an
+ * argument.
+ *
+ * Removed init function, since all initialization is now done in
+ * init_lpc10_encoder_state().
+ *
+ * Changed name of function from lpcenc_ to lpc10_encode, simply to make
+ * all lpc10 functions have more consistent naming with each other.
+ *
+ * Revision 1.1  1996/08/19  22:31:48  jaf
+ * Initial revision
+ * */
+/* Revision 1.3  1996/03/29  22:05:55  jaf */
+/* Commented out the common block variables that are not needed by the */
+/* embedded version. */
+
+/* Revision 1.2  1996/03/26  19:34:50  jaf */
+/* Added comments indicating which constants are not needed in an */
+/* application that uses the LPC-10 coder. */
+
+/* Revision 1.1  1996/02/07  14:44:09  jaf */
+/* Initial revision */
+
+/*   LPC Processing control variables: */
+
+/* *** Read-only: initialized in setup */
+
+/*  Files for Speech, Parameter, and Bitstream Input & Output, */
+/*    and message and debug outputs. */
+
+/* Here are the only files which use these variables: */
+
+/* lpcsim.f setup.f trans.f error.f vqsetup.f */
+
+/* Many files which use fdebug are not listed, since it is only used in */
+/* those other files conditionally, to print trace statements. */
+/* 	integer fsi, fso, fpi, fpo, fbi, fbo, pbin, fmsg, fdebug */
+/*  LPC order, Frame size, Quantization rate, Bits per frame, */
+/*    Error correction */
+/* Subroutine SETUP is the only place where order is assigned a value, */
+/* and that value is 10.  It could increase efficiency 1% or so to */
+/* declare order as a constant (i.e., a Fortran PARAMETER) instead of as 
+*/
+/* a variable in a COMMON block, since it is used in many places in the */
+/* core of the coding and decoding routines.  Actually, I take that back. 
+*/
+/* At least when compiling with f2c, the upper bound of DO loops is */
+/* stored in a local variable before the DO loop begins, and then that is 
+*/
+/* compared against on each iteration. */
+/* Similarly for lframe, which is given a value of MAXFRM in SETUP. */
+/* Similarly for quant, which is given a value of 2400 in SETUP.  quant */
+/* is used in only a few places, and never in the core coding and */
+/* decoding routines, so it could be eliminated entirely. */
+/* nbits is similar to quant, and is given a value of 54 in SETUP. */
+/* corrp is given a value of .TRUE. in SETUP, and is only used in the */
+/* subroutines ENCODE and DECODE.  It doesn't affect the speed of the */
+/* coder significantly whether it is .TRUE. or .FALSE., or whether it is 
+*/
+/* a constant or a variable, since it is only examined once per frame. */
+/* Leaving it as a variable that is set to .TRUE.  seems like a good */
+/* idea, since it does enable some error-correction capability for */
+/* unvoiced frames, with no change in the coding rate, and no noticeable 
+*/
+/* quality difference in the decoded speech. */
+/* 	integer quant, nbits */
+/* *** Read/write: variables for debugging, not needed for LPC algorithm 
+*/
+
+/*  Current frame, Unstable frames, Output clip count, Max onset buffer, 
+*/
+/*    Debug listing detail level, Line count on listing page */
+
+/* nframe is not needed for an embedded LPC10 at all. */
+/* nunsfm is initialized to 0 in SETUP, and incremented in subroutine */
+/* ERROR, which is only called from RCCHK.  When LPC10 is embedded into */
+/* an application, I would recommend removing the call to ERROR in RCCHK, 
+*/
+/* and remove ERROR and nunsfm completely. */
+/* iclip is initialized to 0 in SETUP, and incremented in entry SWRITE in 
+*/
+/* sread.f.  When LPC10 is embedded into an application, one might want */
+/* to cause it to be incremented in a routine that takes the output of */
+/* SYNTHS and sends it to an audio device.  It could be optionally */
+/* displayed, for those that might want to know what it is. */
+/* maxosp is never initialized to 0 in SETUP, although it probably should 
+*/
+/* be, and it is updated in subroutine ANALYS.  I doubt that its value */
+/* would be of much interest to an application in which LPC10 is */
+/* embedded. */
+/* listl and lincnt are not needed for an embedded LPC10 at all. */
+/* 	integer nframe, nunsfm, iclip, maxosp, listl, lincnt */
+/* 	common /contrl/ fsi, fso, fpi, fpo, fbi, fbo, pbin, fmsg, fdebug */
+/* 	common /contrl/ quant, nbits */
+/* 	common /contrl/ nframe, nunsfm, iclip, maxosp, listl, lincnt */
+/*       Local variables that need not be saved */
+/*       Uncoded speech parameters */
+/*       Coded speech parameters */
+/*       Others */
+/*       Local state */
+/*       None */
+    /* Parameter adjustments */
+    if (bits) {
+	--bits;
+	}
+    if (speech) {
+	--speech;
+	}
+
+    /* Function Body */
+
+    chanrd_(&c__10, &ipitv, &irms, irc, &bits[1]);
+    decode_(&ipitv, &irms, irc, voice, &pitch, &rms, rc, st);
+    synths_(voice, &pitch, &rms, rc, &speech[1], &len, st);
+    return 0;
+} /* lpcdec_ */

codecs/lpc10/lpcenc.c

@@ -0,0 +1,163 @@
+/*
+
+$Log$
+Revision 1.4  2000/01/05 08:20:39  markster
+Version 0.1.5 from FTP
+
+Revision 1.2  2000/01/05 08:20:39  markster
+Some OSS fixes and a few lpc changes to make it actually work
+
+ * Revision 1.2  1996/08/20  20:31:21  jaf
+ * Removed all static local variables that were SAVE'd in the Fortran
+ * code, and put them in struct lpc10_encoder_state that is passed as an
+ * argument.
+ *
+ * Removed init function, since all initialization is now done in
+ * init_lpc10_encoder_state().
+ *
+ * Changed name of function from lpcenc_ to lpc10_encode, simply to make
+ * all lpc10 functions have more consistent naming with each other.
+ *
+ * Revision 1.1  1996/08/19  22:31:44  jaf
+ * Initial revision
+ *
+
+*/
+
+#ifdef P_R_O_T_O_T_Y_P_E_S
+extern int lpcenc_(real *speech, integer *bits);
+extern int initlpcenc_(void);
+/*:ref: prepro_ 14 2 6 4 */
+/*:ref: analys_ 14 5 6 4 4 6 6 */
+/*:ref: encode_ 14 7 4 4 6 6 4 4 4 */
+/*:ref: chanwr_ 14 5 4 4 4 4 4 */
+/*:ref: initprepro_ 14 0 */
+/*:ref: initanalys_ 14 0 */
+#endif
+
+/*  -- translated by f2c (version 19951025).
+   You must link the resulting object file with the libraries:
+	-lf2c -lm   (in that order)
+*/
+
+#include "f2c.h"
+
+/* Table of constant values */
+
+static integer c__180 = 180;
+static integer c__10 = 10;
+
+/* ***************************************************************** */
+
+/* $Log$
+ * Revision 1.4  2000/01/05 08:20:39  markster
+ * Version 0.1.5 from FTP
+ *
+/* Revision 1.2  2000/01/05 08:20:39  markster
+/* Some OSS fixes and a few lpc changes to make it actually work
+/*
+ * Revision 1.2  1996/08/20  20:31:21  jaf
+ * Removed all static local variables that were SAVE'd in the Fortran
+ * code, and put them in struct lpc10_encoder_state that is passed as an
+ * argument.
+ *
+ * Removed init function, since all initialization is now done in
+ * init_lpc10_encoder_state().
+ *
+ * Changed name of function from lpcenc_ to lpc10_encode, simply to make
+ * all lpc10 functions have more consistent naming with each other.
+ *
+ * Revision 1.1  1996/08/19  22:31:44  jaf
+ * Initial revision
+ * */
+/* Revision 1.2  1996/03/28  00:01:22  jaf */
+/* Commented out some trace statements. */
+
+/* Revision 1.1  1996/03/28  00:00:27  jaf */
+/* Initial revision */
+
+
+/* ***************************************************************** */
+
+/* Encode one frame of 180 speech samples to 54 bits. */
+
+/* Input: */
+/*  SPEECH - Speech encoded as real values in the range [-1,+1]. */
+/*           Indices 1 through 180 read, and modified (by PREPRO). */
+/* Output: */
+/*  BITS   - 54 encoded bits, stored 1 per array element. */
+/*           Indices 1 through 54 written. */
+
+/* This subroutine maintains local state from one call to the next.  If */
+/* you want to switch to using a new audio stream for this filter, or */
+/* reinitialize its state for any other reason, call the ENTRY */
+/* INITLPCENC. */
+
+/* Subroutine */ int lpc10_encode(real *speech, integer *bits,
+				  struct lpc10_encoder_state *st)
+{
+    integer irms, voice[2], pitch, ipitv;
+    real rc[10];
+    extern /* Subroutine */ int encode_(integer *, integer *, real *, real *, 
+	    integer *, integer *, integer *), chanwr_(integer *, integer *, 
+	    integer *, integer *, integer *, struct lpc10_encoder_state *),
+            analys_(real *, integer *, 
+	    integer *, real *, real *, struct lpc10_encoder_state *),
+            prepro_(real *, integer *, struct lpc10_encoder_state *);
+    integer irc[10];
+    real rms;
+
+/*       Arguments */
+/* $Log$
+ * Revision 1.4  2000/01/05 08:20:39  markster
+ * Version 0.1.5 from FTP
+ *
+/* Revision 1.2  2000/01/05 08:20:39  markster
+/* Some OSS fixes and a few lpc changes to make it actually work
+/*
+ * Revision 1.2  1996/08/20  20:31:21  jaf
+ * Removed all static local variables that were SAVE'd in the Fortran
+ * code, and put them in struct lpc10_encoder_state that is passed as an
+ * argument.
+ *
+ * Removed init function, since all initialization is now done in
+ * init_lpc10_encoder_state().
+ *
+ * Changed name of function from lpcenc_ to lpc10_encode, simply to make
+ * all lpc10 functions have more consistent naming with each other.
+ *
+ * Revision 1.1  1996/08/19  22:31:44  jaf
+ * Initial revision
+ * */
+/* Revision 1.3  1996/03/29  22:03:47  jaf */
+/* Removed definitions for any constants that were no longer used. */
+
+/* Revision 1.2  1996/03/26  19:34:33  jaf */
+/* Added comments indicating which constants are not needed in an */
+/* application that uses the LPC-10 coder. */
+
+/* Revision 1.1  1996/02/07  14:43:51  jaf */
+/* Initial revision */
+
+/*   LPC Configuration parameters: */
+/* Frame size, Prediction order, Pitch period */
+/*       Local variables that need not be saved */
+/*       Uncoded speech parameters */
+/*       Coded speech parameters */
+/*       Local state */
+/*       None */
+    /* Parameter adjustments */
+    if (speech) {
+	--speech;
+	}
+    if (bits) {
+	--bits;
+	}
+
+    /* Function Body */
+    prepro_(&speech[1], &c__180, st);
+    analys_(&speech[1], voice, &pitch, &rms, rc, st);
+    encode_(voice, &pitch, &rms, rc, &ipitv, &irms, irc);
+    chanwr_(&c__10, &ipitv, &irms, irc, &bits[1], st);
+    return 0;
+} /* lpcenc_ */

codecs/lpc10/lpcini.c

@@ -0,0 +1,399 @@
+/*
+
+$Log$
+Revision 1.4  2000/01/05 08:20:39  markster
+Version 0.1.5 from FTP
+
+Revision 1.2  2000/01/05 08:20:39  markster
+Some OSS fixes and a few lpc changes to make it actually work
+
+ * Revision 1.2  1996/08/20  20:35:41  jaf
+ * Added functions for allocating and initializing lpc10_encoder_state
+ * and lpc10_decoder_state structures.
+ *
+ * Revision 1.1  1996/08/19  22:31:40  jaf
+ * Initial revision
+ *
+
+*/
+
+#ifdef P_R_O_T_O_T_Y_P_E_S
+extern int lpcini_(void);
+/* comlen contrl_ 12 */
+/*:ref: initlpcenc_ 14 0 */
+/*:ref: initlpcdec_ 14 0 */
+#endif
+
+/*  -- translated by f2c (version 19951025).
+   You must link the resulting object file with the libraries:
+	-lf2c -lm   (in that order)
+*/
+
+#include "f2c.h"
+
+#include <malloc.h>
+
+/* Common Block Declarations */
+
+struct {
+    integer order, lframe;
+    logical corrp;
+} contrl_;
+
+#define contrl_1 contrl_
+
+/* ***************************************************************** */
+
+/* $Log$
+ * Revision 1.4  2000/01/05 08:20:39  markster
+ * Version 0.1.5 from FTP
+ *
+/* Revision 1.2  2000/01/05 08:20:39  markster
+/* Some OSS fixes and a few lpc changes to make it actually work
+/*
+ * Revision 1.2  1996/08/20  20:35:41  jaf
+ * Added functions for allocating and initializing lpc10_encoder_state
+ * and lpc10_decoder_state structures.
+ *
+ * Revision 1.1  1996/08/19  22:31:40  jaf
+ * Initial revision
+ * */
+/* Revision 1.1  1996/03/28  00:04:05  jaf */
+/* Initial revision */
+
+
+/* ***************************************************************** */
+
+/* Initialize COMMON block variables used by LPC-10 encoder and decoder, */
+/* and call initialization routines for both of them. */
+
+/* Subroutine */ int lpcini_(void)
+{
+
+/* $Log$
+ * Revision 1.4  2000/01/05 08:20:39  markster
+ * Version 0.1.5 from FTP
+ *
+/* Revision 1.2  2000/01/05 08:20:39  markster
+/* Some OSS fixes and a few lpc changes to make it actually work
+/*
+ * Revision 1.2  1996/08/20  20:35:41  jaf
+ * Added functions for allocating and initializing lpc10_encoder_state
+ * and lpc10_decoder_state structures.
+ *
+ * Revision 1.1  1996/08/19  22:31:40  jaf
+ * Initial revision
+ * */
+/* Revision 1.3  1996/03/29  22:03:47  jaf */
+/* Removed definitions for any constants that were no longer used. */
+
+/* Revision 1.2  1996/03/26  19:34:33  jaf */
+/* Added comments indicating which constants are not needed in an */
+/* application that uses the LPC-10 coder. */
+
+/* Revision 1.1  1996/02/07  14:43:51  jaf */
+/* Initial revision */
+
+/*   LPC Configuration parameters: */
+/* Frame size, Prediction order, Pitch period */
+/* $Log$
+ * Revision 1.4  2000/01/05 08:20:39  markster
+ * Version 0.1.5 from FTP
+ *
+/* Revision 1.2  2000/01/05 08:20:39  markster
+/* Some OSS fixes and a few lpc changes to make it actually work
+/*
+ * Revision 1.2  1996/08/20  20:35:41  jaf
+ * Added functions for allocating and initializing lpc10_encoder_state
+ * and lpc10_decoder_state structures.
+ *
+ * Revision 1.1  1996/08/19  22:31:40  jaf
+ * Initial revision
+ * */
+/* Revision 1.3  1996/03/29  22:05:55  jaf */
+/* Commented out the common block variables that are not needed by the */
+/* embedded version. */
+
+/* Revision 1.2  1996/03/26  19:34:50  jaf */
+/* Added comments indicating which constants are not needed in an */
+/* application that uses the LPC-10 coder. */
+
+/* Revision 1.1  1996/02/07  14:44:09  jaf */
+/* Initial revision */
+
+/*   LPC Processing control variables: */
+
+/* *** Read-only: initialized in setup */
+
+/*  Files for Speech, Parameter, and Bitstream Input & Output, */
+/*    and message and debug outputs. */
+
+/* Here are the only files which use these variables: */
+
+/* lpcsim.f setup.f trans.f error.f vqsetup.f */
+
+/* Many files which use fdebug are not listed, since it is only used in */
+/* those other files conditionally, to print trace statements. */
+/* 	integer fsi, fso, fpi, fpo, fbi, fbo, pbin, fmsg, fdebug */
+/*  LPC order, Frame size, Quantization rate, Bits per frame, */
+/*    Error correction */
+/* Subroutine SETUP is the only place where order is assigned a value, */
+/* and that value is 10.  It could increase efficiency 1% or so to */
+/* declare order as a constant (i.e., a Fortran PARAMETER) instead of as 
+*/
+/* a variable in a COMMON block, since it is used in many places in the */
+/* core of the coding and decoding routines.  Actually, I take that back. 
+*/
+/* At least when compiling with f2c, the upper bound of DO loops is */
+/* stored in a local variable before the DO loop begins, and then that is 
+*/
+/* compared against on each iteration. */
+/* Similarly for lframe, which is given a value of MAXFRM in SETUP. */
+/* Similarly for quant, which is given a value of 2400 in SETUP.  quant */
+/* is used in only a few places, and never in the core coding and */
+/* decoding routines, so it could be eliminated entirely. */
+/* nbits is similar to quant, and is given a value of 54 in SETUP. */
+/* corrp is given a value of .TRUE. in SETUP, and is only used in the */
+/* subroutines ENCODE and DECODE.  It doesn't affect the speed of the */
+/* coder significantly whether it is .TRUE. or .FALSE., or whether it is 
+*/
+/* a constant or a variable, since it is only examined once per frame. */
+/* Leaving it as a variable that is set to .TRUE.  seems like a good */
+/* idea, since it does enable some error-correction capability for */
+/* unvoiced frames, with no change in the coding rate, and no noticeable 
+*/
+/* quality difference in the decoded speech. */
+/* 	integer quant, nbits */
+/* *** Read/write: variables for debugging, not needed for LPC algorithm 
+*/
+
+/*  Current frame, Unstable frames, Output clip count, Max onset buffer, 
+*/
+/*    Debug listing detail level, Line count on listing page */
+
+/* nframe is not needed for an embedded LPC10 at all. */
+/* nunsfm is initialized to 0 in SETUP, and incremented in subroutine */
+/* ERROR, which is only called from RCCHK.  When LPC10 is embedded into */
+/* an application, I would recommend removing the call to ERROR in RCCHK, 
+*/
+/* and remove ERROR and nunsfm completely. */
+/* iclip is initialized to 0 in SETUP, and incremented in entry SWRITE in 
+*/
+/* sread.f.  When LPC10 is embedded into an application, one might want */
+/* to cause it to be incremented in a routine that takes the output of */
+/* SYNTHS and sends it to an audio device.  It could be optionally */
+/* displayed, for those that might want to know what it is. */
+/* maxosp is never initialized to 0 in SETUP, although it probably should 
+*/
+/* be, and it is updated in subroutine ANALYS.  I doubt that its value */
+/* would be of much interest to an application in which LPC10 is */
+/* embedded. */
+/* listl and lincnt are not needed for an embedded LPC10 at all. */
+/* 	integer nframe, nunsfm, iclip, maxosp, listl, lincnt */
+/* 	common /contrl/ fsi, fso, fpi, fpo, fbi, fbo, pbin, fmsg, fdebug */
+/* 	common /contrl/ quant, nbits */
+/* 	common /contrl/ nframe, nunsfm, iclip, maxosp, listl, lincnt */
+    contrl_1.order = 10;
+    contrl_1.lframe = 180;
+    contrl_1.corrp = TRUE_;
+    return 0;
+} /* lpcini_ */
+
+
+
+/* Allocate memory for, and initialize, the state that needs to be
+   kept from encoding one frame to the next for a single
+   LPC-10-compressed audio stream.  Return 0 if malloc fails,
+   otherwise return pointer to new structure. */
+
+struct lpc10_encoder_state *
+create_lpc10_encoder_state()
+{
+    struct lpc10_encoder_state *st;
+
+    st = (struct lpc10_encoder_state *)
+	malloc((unsigned) sizeof (struct lpc10_encoder_state));
+    if (st != 0) {
+	init_lpc10_encoder_state(st);
+    }
+    return (st);
+}
+
+
+
+void init_lpc10_encoder_state(struct lpc10_encoder_state *st)
+{
+    int i;
+
+    lpcini_();
+
+    /* State used only by function hp100 */
+    st->z11 = 0.0f;
+    st->z21 = 0.0f;
+    st->z12 = 0.0f;
+    st->z22 = 0.0f;
+    
+    /* State used by function analys */
+    for (i = 0; i < 540; i++) {
+	st->inbuf[i] = 0.0f;
+	st->pebuf[i] = 0.0f;
+    }
+    for (i = 0; i < 696; i++) {
+	st->lpbuf[i] = 0.0f;
+    }
+    for (i = 0; i < 312; i++) {
+	st->ivbuf[i] = 0.0f;
+    }
+    st->bias = 0.0f;
+    /* integer osbuf[10];  /* no initial value necessary */
+    st->osptr = 1;
+    for (i = 0; i < 3; i++) {
+	st->obound[i] = 0;
+    }
+    st->vwin[4] = 307;
+    st->vwin[5] = 462;
+    st->awin[4] = 307;
+    st->awin[5] = 462;
+    for (i = 0; i < 8; i++) {
+	st->voibuf[i] = 0;
+    }
+    for (i = 0; i < 3; i++) {
+	st->rmsbuf[i] = 0.0f;
+    }
+    for (i = 0; i < 30; i++) {
+	st->rcbuf[i] = 0.0f;
+    }
+    st->zpre = 0.0f;
+
+
+    /* State used by function onset */
+    st->n = 0.0f;
+    st->d__ = 1.0f;
+    /* real fpc;   /* no initial value necessary */
+    for (i = 0; i < 16; i++) {
+	st->l2buf[i] = 0.0f;
+    }
+    st->l2sum1 = 0.0f;
+    st->l2ptr1 = 1;
+    st->l2ptr2 = 9;
+    /* integer lasti;    /* no initial value necessary */
+    st->hyst = FALSE_;
+
+    /* State used by function voicin */
+    st->dither = 20.0f;
+    st->maxmin = 0.0f;
+    for (i = 0; i < 6; i++) {
+	st->voice[i] = 0.0f;
+    }
+    st->lbve = 3000;
+    st->fbve = 3000;
+    st->fbue = 187;
+    st->ofbue = 187;
+    st->sfbue = 187;
+    st->lbue = 93;
+    st->olbue = 93;
+    st->slbue = 93;
+    st->snr = (real) (st->fbve / st->fbue << 6);
+
+    /* State used by function dyptrk */
+    for (i = 0; i < 60; i++) {
+	st->s[i] = 0.0f;
+    }
+    for (i = 0; i < 120; i++) {
+	st->p[i] = 0;
+    }
+    st->ipoint = 0;
+    st->alphax = 0.0f;
+
+    /* State used by function chanwr */
+    st->isync = 0;
+
+}
+
+
+
+/* Allocate memory for, and initialize, the state that needs to be
+   kept from decoding one frame to the next for a single
+   LPC-10-compressed audio stream.  Return 0 if malloc fails,
+   otherwise return pointer to new structure. */
+
+struct lpc10_decoder_state *
+create_lpc10_decoder_state()
+{
+    struct lpc10_decoder_state *st;
+
+    st = (struct lpc10_decoder_state *)
+	malloc((unsigned) sizeof (struct lpc10_decoder_state));
+    if (st != 0) {
+	init_lpc10_decoder_state(st);
+    }
+    return (st);
+}
+
+
+
+void init_lpc10_decoder_state(struct lpc10_decoder_state *st)
+{
+    int i;
+
+    lpcini_();
+
+    /* State used by function decode */
+    st->iptold = 60;
+    st->first = TRUE_;
+    st->ivp2h = 0;
+    st->iovoic = 0;
+    st->iavgp = 60;
+    st->erate = 0;
+    for (i = 0; i < 30; i++) {
+	st->drc[i] = 0;
+    }
+    for (i = 0; i < 3; i++) {
+	st->dpit[i] = 0;
+	st->drms[i] = 0;
+    }
+
+    /* State used by function synths */
+    for (i = 0; i < 360; i++) {
+	st->buf[i] = 0.0f;
+    }
+    st->buflen = 180;
+
+    /* State used by function pitsyn */
+    /* ivoico;   /* no initial value necessary as long as first_pitsyn is initially TRUE_ */
+    /* ipito;   /* no initial value necessary as long as first_pitsyn is initially TRUE_ */
+    st->rmso = 1.0f;
+    /* rco[10];   /* no initial value necessary as long as first_pitsyn is initially TRUE_ */
+    /* integer jsamp;   /* no initial value necessary as long as first_pitsyn is initially TRUE_ */
+    st->first_pitsyn = TRUE_;
+
+    /* State used by function bsynz */
+    st->ipo = 0;
+    for (i = 0; i < 166; i++) {
+	st->exc[i] = 0.0f;
+	st->exc2[i] = 0.0f;
+    }
+    st->lpi1 = 0.0f;
+    st->lpi2 = 0.0f;
+    st->lpi3 = 0.0f;
+    st->hpi1 = 0.0f;
+    st->hpi2 = 0.0f;
+    st->hpi3 = 0.0f;
+    st->rmso_bsynz = 0.0f;
+
+    /* State used by function random */
+    st->j = 2;
+    st->k = 5;
+    st->y[0] = (shortint) -21161;
+    st->y[1] = (shortint) -8478;
+    st->y[2] = (shortint) 30892;
+    st->y[3] = (shortint) -10216;
+    st->y[4] = (shortint) 16950;
+
+    /* State used by function deemp */
+    st->dei1 = 0.0f;
+    st->dei2 = 0.0f;
+    st->deo1 = 0.0f;
+    st->deo2 = 0.0f;
+    st->deo3 = 0.0f;
+}

codecs/lpc10/lpfilt.c

@@ -0,0 +1,113 @@
+/*
+
+$Log$
+Revision 1.4  2000/01/05 08:20:39  markster
+Version 0.1.5 from FTP
+
+Revision 1.2  2000/01/05 08:20:39  markster
+Some OSS fixes and a few lpc changes to make it actually work
+
+ * Revision 1.1  1996/08/19  22:31:35  jaf
+ * Initial revision
+ *
+
+*/
+
+#ifdef P_R_O_T_O_T_Y_P_E_S
+extern int lpfilt_(real *inbuf, real *lpbuf, integer *len, integer *nsamp);
+#endif
+
+/*  -- translated by f2c (version 19951025).
+   You must link the resulting object file with the libraries:
+	-lf2c -lm   (in that order)
+*/
+
+#include "f2c.h"
+
+/* *********************************************************************** */
+
+/* 	LPFILT Version 55 */
+
+/* $Log$
+ * Revision 1.4  2000/01/05 08:20:39  markster
+ * Version 0.1.5 from FTP
+ *
+/* Revision 1.2  2000/01/05 08:20:39  markster
+/* Some OSS fixes and a few lpc changes to make it actually work
+/*
+ * Revision 1.1  1996/08/19  22:31:35  jaf
+ * Initial revision
+ * */
+/* Revision 1.3  1996/03/15  16:53:49  jaf */
+/* Just put comment header in standard form. */
+
+/* Revision 1.2  1996/03/12  23:58:06  jaf */
+/* Comments added explaining that none of the local variables of this */
+/* subroutine need to be saved from one invocation to the next. */
+
+/* Revision 1.1  1996/02/07 14:47:44  jaf */
+/* Initial revision */
+
+
+/* *********************************************************************** */
+
+/*   31 Point Equiripple FIR Low-Pass Filter */
+/*     Linear phase, delay = 15 samples */
+
+/* 	Passband:  ripple = 0.25 dB, cutoff =  800 Hz */
+/* 	Stopband:  atten. =  40. dB, cutoff = 1240 Hz */
+
+/* Inputs: */
+/*  LEN    - Length of speech buffers */
+/*  NSAMP  - Number of samples to filter */
+/*  INBUF  - Input speech buffer */
+/*           Indices len-nsamp-29 through len are read. */
+/* Output: */
+/*  LPBUF  - Low passed speech buffer (must be different array than INBUF) */
+/*           Indices len+1-nsamp through len are written. */
+
+/* This subroutine has no local state. */
+
+/* Subroutine */ int lpfilt_(real *inbuf, real *lpbuf, integer *len, integer *
+	nsamp)
+{
+    /* System generated locals */
+    integer i__1;
+
+    /* Local variables */
+    integer j;
+    real t;
+
+/* 	Arguments */
+/* 	Parameters/constants */
+/*       Local variables that need not be saved */
+/*       Local state */
+/*       None */
+    /* Parameter adjustments */
+    --lpbuf;
+    --inbuf;
+
+    /* Function Body */
+    i__1 = *len;
+    for (j = *len + 1 - *nsamp; j <= i__1; ++j) {
+	t = (inbuf[j] + inbuf[j - 30]) * -.0097201988f;
+	t += (inbuf[j - 1] + inbuf[j - 29]) * -.0105179986f;
+	t += (inbuf[j - 2] + inbuf[j - 28]) * -.0083479648f;
+	t += (inbuf[j - 3] + inbuf[j - 27]) * 5.860774e-4f;
+	t += (inbuf[j - 4] + inbuf[j - 26]) * .0130892089f;
+	t += (inbuf[j - 5] + inbuf[j - 25]) * .0217052232f;
+	t += (inbuf[j - 6] + inbuf[j - 24]) * .0184161253f;
+	t += (inbuf[j - 7] + inbuf[j - 23]) * 3.39723e-4f;
+	t += (inbuf[j - 8] + inbuf[j - 22]) * -.0260797087f;
+	t += (inbuf[j - 9] + inbuf[j - 21]) * -.0455563702f;
+	t += (inbuf[j - 10] + inbuf[j - 20]) * -.040306855f;
+	t += (inbuf[j - 11] + inbuf[j - 19]) * 5.029835e-4f;
+	t += (inbuf[j - 12] + inbuf[j - 18]) * .0729262903f;
+	t += (inbuf[j - 13] + inbuf[j - 17]) * .1572008878f;
+	t += (inbuf[j - 14] + inbuf[j - 16]) * .2247288674f;
+	t += inbuf[j - 15] * .250535965f;
+	lpbuf[j] = t;
+    }
+    return 0;
+} /* lpfilt_ */
+

codecs/lpc10/median.c

@@ -0,0 +1,77 @@
+/*
+
+$Log$
+Revision 1.4  2000/01/05 08:20:39  markster
+Version 0.1.5 from FTP
+
+Revision 1.2  2000/01/05 08:20:39  markster
+Some OSS fixes and a few lpc changes to make it actually work
+
+ * Revision 1.1  1996/08/19  22:31:31  jaf
+ * Initial revision
+ *
+
+*/
+
+#ifdef P_R_O_T_O_T_Y_P_E_S
+extern integer median_(integer *d1, integer *d2, integer *d3);
+#endif
+
+/*  -- translated by f2c (version 19951025).
+   You must link the resulting object file with the libraries:
+	-lf2c -lm   (in that order)
+*/
+
+#include "f2c.h"
+
+/* ********************************************************************* */
+
+/* 	MEDIAN Version 45G */
+
+/* $Log$
+ * Revision 1.4  2000/01/05 08:20:39  markster
+ * Version 0.1.5 from FTP
+ *
+/* Revision 1.2  2000/01/05 08:20:39  markster
+/* Some OSS fixes and a few lpc changes to make it actually work
+/*
+ * Revision 1.1  1996/08/19  22:31:31  jaf
+ * Initial revision
+ * */
+/* Revision 1.2  1996/03/14  22:30:22  jaf */
+/* Just rearranged the comments and local variable declarations a bit. */
+
+/* Revision 1.1  1996/02/07 14:47:53  jaf */
+/* Initial revision */
+
+
+/* ********************************************************************* */
+
+/*  Find median of three values */
+
+/* Input: */
+/*  D1,D2,D3 - Three input values */
+/* Output: */
+/*  MEDIAN - Median value */
+
+integer median_(integer *d1, integer *d2, integer *d3)
+{
+    /* System generated locals */
+    integer ret_val;
+
+/*       Arguments */
+    ret_val = *d2;
+    if (*d2 > *d1 && *d2 > *d3) {
+	ret_val = *d1;
+	if (*d3 > *d1) {
+	    ret_val = *d3;
+	}
+    } else if (*d2 < *d1 && *d2 < *d3) {
+	ret_val = *d1;
+	if (*d3 < *d1) {
+	    ret_val = *d3;
+	}
+    }
+    return ret_val;
+} /* median_ */
+

codecs/lpc10/mload.c

@@ -0,0 +1,151 @@
+/*
+
+$Log$
+Revision 1.4  2000/01/05 08:20:39  markster
+Version 0.1.5 from FTP
+
+Revision 1.2  2000/01/05 08:20:39  markster
+Some OSS fixes and a few lpc changes to make it actually work
+
+ * Revision 1.1  1996/08/19  22:31:25  jaf
+ * Initial revision
+ *
+
+*/
+
+#ifdef P_R_O_T_O_T_Y_P_E_S
+extern int mload_(integer *order, integer *awins, integer *awinf, real *speech, real *phi, real *psi);
+#endif
+
+/*  -- translated by f2c (version 19951025).
+   You must link the resulting object file with the libraries:
+	-lf2c -lm   (in that order)
+*/
+
+#include "f2c.h"
+
+/* ***************************************************************** */
+
+/* 	MLOAD Version 48 */
+
+/* $Log$
+ * Revision 1.4  2000/01/05 08:20:39  markster
+ * Version 0.1.5 from FTP
+ *
+/* Revision 1.2  2000/01/05 08:20:39  markster
+/* Some OSS fixes and a few lpc changes to make it actually work
+/*
+ * Revision 1.1  1996/08/19  22:31:25  jaf
+ * Initial revision
+ * */
+/* Revision 1.5  1996/03/27  23:59:51  jaf */
+/* Added some more accurate comments about which indices of the argument */
+/* array SPEECH are read.  I thought that this might be the cause of a */
+/* problem I've been having, but it isn't. */
+
+/* Revision 1.4  1996/03/26  19:16:53  jaf */
+/* Commented out the code at the end that copied the lower triangular */
+/* half of PHI into the upper triangular half (making the resulting */
+/* matrix symmetric).  The upper triangular half was never used by later */
+/* code in subroutine ANALYS. */
+
+/* Revision 1.3  1996/03/18  21:16:00  jaf */
+/* Just added a few comments about which array indices of the arguments */
+/* are used, and mentioning that this subroutine has no local state. */
+
+/* Revision 1.2  1996/03/13  16:47:41  jaf */
+/* Comments added explaining that none of the local variables of this */
+/* subroutine need to be saved from one invocation to the next. */
+
+/* Revision 1.1  1996/02/07 14:48:01  jaf */
+/* Initial revision */
+
+
+/* ***************************************************************** */
+
+/* Load a covariance matrix. */
+
+/* Input: */
+/*  ORDER            - Analysis order */
+/*  AWINS            - Analysis window start */
+/*  AWINF            - Analysis window finish */
+/*  SPEECH(AWINF)    - Speech buffer */
+/*                     Indices MIN(AWINS, AWINF-(ORDER-1)) through */
+/*                             MAX(AWINF, AWINS+(ORDER-1)) read. */
+/*                     As long as (AWINF-AWINS) .GE. (ORDER-1), */
+/*                     this is just indices AWINS through AWINF. */
+/* Output: */
+/*  PHI(ORDER,ORDER) - Covariance matrix */
+/*                    Lower triangular half and diagonal written, and read.*/
+/*                     Upper triangular half untouched. */
+/*  PSI(ORDER)       - Prediction vector */
+/*                     Indices 1 through ORDER written, */
+/*                     and most are read after that. */
+
+/* This subroutine has no local state. */
+
+/* Subroutine */ int mload_(integer *order, integer *awins, integer *awinf, 
+	real *speech, real *phi, real *psi)
+{
+    /* System generated locals */
+    integer phi_dim1, phi_offset, i__1, i__2;
+
+    /* Local variables */
+    integer c__, i__, r__, start;
+
+/*       Arguments */
+/*       Local variables that need not be saved */
+/*   Load first column of triangular covariance matrix PHI */
+    /* Parameter adjustments */
+    --psi;
+    phi_dim1 = *order;
+    phi_offset = phi_dim1 + 1;
+    phi -= phi_offset;
+    --speech;
+
+    /* Function Body */
+    start = *awins + *order;
+    i__1 = *order;
+    for (r__ = 1; r__ <= i__1; ++r__) {
+	phi[r__ + phi_dim1] = 0.f;
+	i__2 = *awinf;
+	for (i__ = start; i__ <= i__2; ++i__) {
+	    phi[r__ + phi_dim1] += speech[i__ - 1] * speech[i__ - r__];
+	}
+    }
+/*   Load last element of vector PSI */
+    psi[*order] = 0.f;
+    i__1 = *awinf;
+    for (i__ = start; i__ <= i__1; ++i__) {
+	psi[*order] += speech[i__] * speech[i__ - *order];
+    }
+/*   End correct to get additional columns of PHI */
+    i__1 = *order;
+    for (r__ = 2; r__ <= i__1; ++r__) {
+	i__2 = r__;
+	for (c__ = 2; c__ <= i__2; ++c__) {
+	    phi[r__ + c__ * phi_dim1] = phi[r__ - 1 + (c__ - 1) * phi_dim1] - 
+		    speech[*awinf + 1 - r__] * speech[*awinf + 1 - c__] + 
+		    speech[start - r__] * speech[start - c__];
+	}
+    }
+/*   End correct to get additional elements of PSI */
+    i__1 = *order - 1;
+    for (c__ = 1; c__ <= i__1; ++c__) {
+	psi[c__] = phi[c__ + 1 + phi_dim1] - speech[start - 1] * speech[start 
+		- 1 - c__] + speech[*awinf] * speech[*awinf - c__];
+    }
+/*   Copy lower triangular section into upper (why bother?) */
+/*       I'm commenting this out, since the upper triangular half of PHI 
+*/
+/*       is never used by later code, unless a sufficiently high level of 
+*/
+/*       tracing is turned on. */
+/* 	DO R = 1,ORDER */
+/* 	   DO C = 1,R-1 */
+/* 	      PHI(C,R) = PHI(R,C) */
+/* 	   END DO */
+/* 	END DO */
+    return 0;
+} /* mload_ */
+

codecs/lpc10/onset.c

@@ -0,0 +1,306 @@
+/*
+
+$Log$
+Revision 1.4  2000/01/05 08:20:39  markster
+Version 0.1.5 from FTP
+
+Revision 1.2  2000/01/05 08:20:39  markster
+Some OSS fixes and a few lpc changes to make it actually work
+
+ * Revision 1.2  1996/08/20  20:37:55  jaf
+ * Removed all static local variables that were SAVE'd in the Fortran
+ * code, and put them in struct lpc10_encoder_state that is passed as an
+ * argument.
+ *
+ * Removed init function, since all initialization is now done in
+ * init_lpc10_encoder_state().
+ *
+ * Revision 1.1  1996/08/19  22:31:18  jaf
+ * Initial revision
+ *
+
+*/
+
+#ifdef P_R_O_T_O_T_Y_P_E_S
+extern int onset_(real *pebuf, integer *osbuf, integer *osptr, integer *oslen, integer *sbufl, integer *sbufh, integer *lframe, struct lpc10_encoder_state *st);
+#endif
+
+/*  -- translated by f2c (version 19951025).
+   You must link the resulting object file with the libraries:
+	-lf2c -lm   (in that order)
+*/
+
+#include "f2c.h"
+
+/* Table of constant values */
+
+static real c_b2 = 1.f;
+
+/* ****************************************************************** */
+
+/* 	ONSET Version 49 */
+
+/* $Log$
+ * Revision 1.4  2000/01/05 08:20:39  markster
+ * Version 0.1.5 from FTP
+ *
+/* Revision 1.2  2000/01/05 08:20:39  markster
+/* Some OSS fixes and a few lpc changes to make it actually work
+/*
+ * Revision 1.2  1996/08/20  20:37:55  jaf
+ * Removed all static local variables that were SAVE'd in the Fortran
+ * code, and put them in struct lpc10_encoder_state that is passed as an
+ * argument.
+ *
+ * Removed init function, since all initialization is now done in
+ * init_lpc10_encoder_state().
+ *
+ * Revision 1.1  1996/08/19  22:31:18  jaf
+ * Initial revision
+ * */
+/* Revision 1.5  1996/03/15  16:41:01  jaf */
+/* Just rearranged INITONSET assignment orders to be consistent with */
+/* order of DATA statements in ONSET. */
+
+/* Revision 1.4  1996/03/15  15:48:27  jaf */
+/* Changed some comments, and only reordered the DATA statements (their */
+/* meaning wasn't changed). */
+
+/* Revision 1.3  1996/03/14  23:53:06  jaf */
+/* Added an entry INITONSET that reinitializes the local state variables */
+/* of subroutine ONSET. */
+
+/* Rearranged quite a few comments, adding more explaining which */
+/* arguments were inputs, and how the modified ones can be changed. */
+
+/* Revision 1.2  1996/03/12  23:53:00  jaf */
+/* Lots of comments added about the local state of this subroutine that */
+/* must be saved from one invocation to the next. */
+
+/* One constant 180 replaced with LFRAME, which should be "more general", */
+/* even though it would probably require many more changes than this to */
+/* get this coder to work for other frame sizes. */
+
+/* Revision 1.1  1996/02/07 14:48:09  jaf */
+/* Initial revision */
+
+
+/* ****************************************************************** */
+
+/* 	Floating point version */
+
+
+/*   Detection of onsets in (or slightly preceding) the futuremost frame */
+/*   of speech. */
+
+
+/* Input: */
+/*  PEBUF(SBUFL:SBUFH)  - Preemphasized speech */
+/*                        Indices SBUFH-LFRAME through SBUFH are read. */
+/*  OSLEN  - Maximum number of onsets that can be stored in OSBUF. */
+/*  SBUFL, SBUFH        - Range of PEBUF */
+/*  LFRAME              - length of a frame, in samples */
+/* Input/Output: */
+/*  OSBUF(OSLEN) - Buffer which holds sorted indexes of onsets */
+/*                 Indices A through B are modified, where A */
+/*                 is the original value of OSPTR, and B is the final */
+/*                 value of OSPTR-1.  B is at most OSLEN. */
+/*  OSPTR        - Free pointer into OSBUF */
+/*                 Initial value should be .LE. OSLEN+1. */
+/*                 If so, final value grows by one for each new onset */
+/*                 found, and final value will be .LE. OSLEN+1. */
+
+/* This subroutine maintains local state from one call to the next.  If */
+/* you want to switch to using a new audio stream for this subroutine, or */
+/* reinitialize its state for any other reason, call the ENTRY INITONSET. */
+
+/* Subroutine */ int onset_(real *pebuf, integer *osbuf, integer *
+	osptr, integer *oslen, integer *sbufl, integer *sbufh, integer *
+	lframe, struct lpc10_encoder_state *st)
+{
+    /* Initialized data */
+
+    real *n;
+    real *d__;
+    real *l2buf;
+    real *l2sum1;
+    integer *l2ptr1;
+    integer *l2ptr2;
+    logical *hyst;
+
+    /* System generated locals */
+    integer pebuf_offset, i__1;
+    real r__1;
+
+    /* Builtin functions */
+    double r_sign(real *, real *);
+
+    /* Local variables */
+    integer i__;
+    integer *lasti;
+    real l2sum2;
+    real *fpc;
+
+/*       Arguments */
+/* $Log$
+ * Revision 1.4  2000/01/05 08:20:39  markster
+ * Version 0.1.5 from FTP
+ *
+/* Revision 1.2  2000/01/05 08:20:39  markster
+/* Some OSS fixes and a few lpc changes to make it actually work
+/*
+ * Revision 1.2  1996/08/20  20:37:55  jaf
+ * Removed all static local variables that were SAVE'd in the Fortran
+ * code, and put them in struct lpc10_encoder_state that is passed as an
+ * argument.
+ *
+ * Removed init function, since all initialization is now done in
+ * init_lpc10_encoder_state().
+ *
+ * Revision 1.1  1996/08/19  22:31:18  jaf
+ * Initial revision
+ * */
+/* Revision 1.3  1996/03/29  22:03:47  jaf */
+/* Removed definitions for any constants that were no longer used. */
+
+/* Revision 1.2  1996/03/26  19:34:33  jaf */
+/* Added comments indicating which constants are not needed in an */
+/* application that uses the LPC-10 coder. */
+
+/* Revision 1.1  1996/02/07  14:43:51  jaf */
+/* Initial revision */
+
+/*   LPC Configuration parameters: */
+/* Frame size, Prediction order, Pitch period */
+/*       Parameters/constants */
+/*   Parameters for onset detection algorithm: */
+/*    L2		Threshold for filtered slope of FPC (function of L2WID!) */
+/*    L2LAG	Lag due to both filters which compute filtered slope of FPC */
+/*    L2WID	Width of the filter which computes the slope of FPC */
+/*    OSHYST	The number of samples of slope(FPC) which must be below */
+/* 	        the threshold before a new onset may be declared. */
+/*       Local variables that need not be saved */
+/*       Local state */
+/*   Variables */
+/*    N, D       Numerator and denominator of prediction filters */
+/*    FPC        Current prediction coefs */
+/*    L2BUF, L2SUM1, L2SUM2    State of slope filter */
+/*       The only "significant" change I've made is to change L2SUM2 out 
+*/
+/*       of the list of local variables that need to be saved, since it */
+/*       didn't need to be. */
+/*       L2SUM1 need not be, but avoiding saving it would require a small 
+*/
+/*       change to the body of the code.  See comments below for an */
+/*       example of how the code could be changed to avoid saving L2SUM1. 
+*/
+/*       FPC and LASTI are saved from one invocation to the next, but */
+/*       they are not given initial values.  This is acceptable, because 
+*/
+/*       FPC will be assigned a value the first time that this function */
+/*       is called after D is initialized to 1, since the formula to */
+/*       change D will not change it to 0 in one step, and the IF (D */
+/*       .NE. 0) statement will execute its THEN part, initializing FPC. 
+*/
+
+/*       LASTI's value will not be used until HYST is .TRUE., and */
+/*       whenever HYST is changed from its initial value of .FALSE., */
+/*       LASTI is assigned a value. */
+/*       In a C version of this coder, it would be nice if all of these */
+/*       saved things, in this and all other subroutines, could be stored 
+*/
+/*       in a single struct lpc10_coder_state_t, initialized with a call 
+*/
+/*       to a function like lpc10_init(&lpc10_coder_state).  In this way, 
+*/
+/*       a program that used these functions could conveniently alternate 
+*/
+/*       coding more than one distinct audio stream. */
+
+    n = &(st->n);
+    d__ = &(st->d__);
+    fpc = &(st->fpc);
+    l2buf = &(st->l2buf[0]);
+    l2sum1 = &(st->l2sum1);
+    l2ptr1 = &(st->l2ptr1);
+    l2ptr2 = &(st->l2ptr2);
+    lasti = &(st->lasti);
+    hyst = &(st->hyst);
+
+    /* Parameter adjustments */
+    if (osbuf) {
+	--osbuf;
+	}
+    if (pebuf) {
+	pebuf_offset = *sbufl;
+	pebuf -= pebuf_offset;
+	}
+
+    /* Function Body */
+
+/*       The following line subtracted a hard-coded "180" from LASTI, */
+/*       instead of using a variable like LFRAME or a constant like */
+/*       MAXFRM.  I changed it to LFRAME, for "generality". */
+    if (*hyst) {
+	*lasti -= *lframe;
+    }
+    i__1 = *sbufh;
+    for (i__ = *sbufh - *lframe + 1; i__ <= i__1; ++i__) {
+/*   Compute FPC; Use old FPC on divide by zero; Clamp FPC to +/- 1. 
+*/
+	*n = (pebuf[i__] * pebuf[i__ - 1] + (*n) * 63.f) / 64.f;
+/* Computing 2nd power */
+	r__1 = pebuf[i__ - 1];
+	*d__ = (r__1 * r__1 + (*d__) * 63.f) / 64.f;
+	if ((*d__) != 0.f) {
+	    if (abs(*n) > (*d__)) {
+		*fpc = r_sign(&c_b2, n);
+	    } else {
+		*fpc = (*n) / (*d__);
+	    }
+	}
+/*   Filter FPC */
+/*       In order to allow L2SUM1 not to be saved from one invocation 
+of */
+/*       this subroutine to the next, one could change the sequence of
+ */
+/*       assignments below, up to the IF statement, to the following. 
+ In */
+/*       addition, the initial value of L2PTR2 should be changed to */
+/*       L2WID/2 instead of L2WID/2+1. */
+
+/*       L2SUM1 = L2BUF(L2PTR2) */
+/*       L2PTR2 = MOD(L2PTR2,L2WID)+1 */
+/*       L2SUM1 = L2SUM1 - L2BUF(L2PTR2) + FPC */
+/*       L2BUF(L2PTR2) = L2SUM1 */
+
+/* *       The following lines didn't change from the original: */
+/*       L2SUM2 = L2BUF(L2PTR1) */
+/*       L2BUF(L2PTR1) = FPC */
+/*       L2PTR1 = MOD(L2PTR1,L2WID)+1 */
+
+	l2sum2 = l2buf[*l2ptr1 - 1];
+	*l2sum1 = *l2sum1 - l2buf[*l2ptr2 - 1] + *fpc;
+	l2buf[*l2ptr2 - 1] = *l2sum1;
+	l2buf[*l2ptr1 - 1] = *fpc;
+	*l2ptr1 = *l2ptr1 % 16 + 1;
+	*l2ptr2 = *l2ptr2 % 16 + 1;
+	if ((r__1 = *l2sum1 - l2sum2, abs(r__1)) > 1.7f) {
+	    if (! (*hyst)) {
+/*   Ignore if buffer full */
+		if (*osptr <= *oslen) {
+		    osbuf[*osptr] = i__ - 9;
+		    ++(*osptr);
+		}
+		*hyst = TRUE_;
+	    }
+	    *lasti = i__;
+/*       After one onset detection, at least OSHYST sample times m
+ust go */
+/*       by before another is allowed to occur. */
+	} else if ((*hyst) && i__ - *lasti >= 10) {
+	    *hyst = FALSE_;
+	}
+    }
+    return 0;
+} /* onset_ */

codecs/lpc10/pitsyn.c

@@ -0,0 +1,556 @@
+/*
+
+$Log$
+Revision 1.4  2000/01/05 08:20:39  markster
+Version 0.1.5 from FTP
+
+Revision 1.2  2000/01/05 08:20:39  markster
+Some OSS fixes and a few lpc changes to make it actually work
+
+ * Revision 1.2  1996/08/20  20:40:12  jaf
+ * Removed all static local variables that were SAVE'd in the Fortran
+ * code, and put them in struct lpc10_decoder_state that is passed as an
+ * argument.
+ *
+ * Removed init function, since all initialization is now done in
+ * init_lpc10_decoder_state().
+ *
+ * Revision 1.1  1996/08/19  22:31:12  jaf
+ * Initial revision
+ *
+
+*/
+
+#ifdef P_R_O_T_O_T_Y_P_E_S
+extern int pitsyn_(integer *order, integer *voice, integer *pitch, real *rms, real *rc, integer *lframe, integer *ivuv, integer *ipiti, real *rmsi, real *rci, integer *nout, real *ratio, struct lpc10_decoder_state *st);
+#endif
+
+/*  -- translated by f2c (version 19951025).
+   You must link the resulting object file with the libraries:
+	-lf2c -lm   (in that order)
+*/
+
+#include "f2c.h"
+
+/* ***************************************************************** */
+
+/* 	PITSYN Version 53 */
+
+/* $Log$
+ * Revision 1.4  2000/01/05 08:20:39  markster
+ * Version 0.1.5 from FTP
+ *
+/* Revision 1.2  2000/01/05 08:20:39  markster
+/* Some OSS fixes and a few lpc changes to make it actually work
+/*
+ * Revision 1.2  1996/08/20  20:40:12  jaf
+ * Removed all static local variables that were SAVE'd in the Fortran
+ * code, and put them in struct lpc10_decoder_state that is passed as an
+ * argument.
+ *
+ * Removed init function, since all initialization is now done in
+ * init_lpc10_decoder_state().
+ *
+ * Revision 1.1  1996/08/19  22:31:12  jaf
+ * Initial revision
+ * */
+/* Revision 1.2  1996/03/25  18:49:07  jaf */
+/* Added commments about which indices of array arguments are read or */
+/* written. */
+
+/* Rearranged local variable declarations to indicate which need to be */
+/* saved from one invocation to the next.  Added entry INITPITSYN to */
+/* reinitialize local state variables, if desired. */
+
+/* Added lots of comments about proving that the maximum number of pitch */
+/* periods (NOUT) that can be returned is 16.  The call to STOP that */
+/* could happen if NOUT got too large was removed as a result. */
+
+/* Also proved that the total number of samples returned from N calls, */
+/* each with identical values of LFRAME, will always be in the range */
+/* N*LFRAME-MAXPIT+1 to N*LFRAME. */
+
+/* Revision 1.1  1996/02/07 14:48:18  jaf */
+/* Initial revision */
+
+
+/* ***************************************************************** */
+
+/*   Synthesize a single pitch epoch */
+
+/* Input: */
+/*  ORDER  - Synthesis order (number of RC's) */
+/*  VOICE  - Half frame voicing decisions */
+/*           Indices 1 through 2 read. */
+/*  LFRAME - Length of speech buffer */
+/* Input/Output: */
+/*  PITCH  - Pitch */
+/*           This value should be in the range MINPIT (20) to MAXPIT */
+/*           (156), inclusive. */
+/*           PITCH can be modified under some conditions. */
+/*  RMS    - Energy  (can be modified) */
+/*           RMS is changed to 1 if the value passed in is less than 1. */
+/*  RC     - Reflection coefficients */
+/*           Indices 1 through ORDER can be temporarily overwritten with */
+/*           RCO, and then replaced with original values, under some */
+/*           conditions. */
+/* Output: */
+/*  IVUV   - Pitch epoch voicing decisions */
+/*           Indices (I) of IVUV, IPITI, and RMSI are written, */
+/*           and indices (J,I) of RCI are written, */
+/*           where I ranges from 1 to NOUT, and J ranges from 1 to ORDER. */
+/*  IPITI  - Pitch epoch length */
+/*  RMSI   - Pitch epoch energy */
+/*  RCI    - Pitch epoch RC's */
+/*  NOUT   - Number of pitch periods in this frame */
+/*           This is at least 0, at least 1 if MAXPIT .LT. LFRAME (this */
+/*           is currently true on every call), and can never be more than */
+/*           (LFRAME+MAXPIT-1)/PITCH, which is currently 16 with */
+/*           LFRAME=180, MAXPIT=156, and PITCH .GE. 20, as SYNTHS */
+/*           guarantees when it calls this subroutine. */
+/*  RATIO  - Previous to present energy ratio */
+/*           Always assigned a value. */
+
+/* Subroutine */ int pitsyn_(integer *order, integer *voice, 
+	integer *pitch, real *rms, real *rc, integer *lframe, integer *ivuv, 
+	integer *ipiti, real *rmsi, real *rci, integer *nout, real *ratio,
+			       struct lpc10_decoder_state *st)
+{
+    /* Initialized data */
+
+    real *rmso;
+    logical *first;
+
+    /* System generated locals */
+    integer rci_dim1, rci_offset, i__1, i__2;
+    real r__1;
+
+    /* Builtin functions */
+    double log(doublereal), exp(doublereal);
+
+    /* Local variables */
+    real alrn, alro, yarc[10], prop;
+    integer i__, j, vflag, jused, lsamp;
+    integer *jsamp;
+    real slope;
+    integer *ipito;
+    real uvpit;
+    integer ip, nl, ivoice;
+    integer *ivoico;
+    integer istart;
+    real *rco;
+    real xxy;
+
+/*       Arguments */
+/* $Log$
+ * Revision 1.4  2000/01/05 08:20:39  markster
+ * Version 0.1.5 from FTP
+ *
+/* Revision 1.2  2000/01/05 08:20:39  markster
+/* Some OSS fixes and a few lpc changes to make it actually work
+/*
+ * Revision 1.2  1996/08/20  20:40:12  jaf
+ * Removed all static local variables that were SAVE'd in the Fortran
+ * code, and put them in struct lpc10_decoder_state that is passed as an
+ * argument.
+ *
+ * Removed init function, since all initialization is now done in
+ * init_lpc10_decoder_state().
+ *
+ * Revision 1.1  1996/08/19  22:31:12  jaf
+ * Initial revision
+ * */
+/* Revision 1.3  1996/03/29  22:03:47  jaf */
+/* Removed definitions for any constants that were no longer used. */
+
+/* Revision 1.2  1996/03/26  19:34:33  jaf */
+/* Added comments indicating which constants are not needed in an */
+/* application that uses the LPC-10 coder. */
+
+/* Revision 1.1  1996/02/07  14:43:51  jaf */
+/* Initial revision */
+
+/*   LPC Configuration parameters: */
+/* Frame size, Prediction order, Pitch period */
+/*       Local variables that need not be saved */
+/*       LSAMP is initialized in the IF (FIRST) THEN clause, but it is */
+/*       not used the first time through, and it is given a value before 
+*/
+/*       use whenever FIRST is .FALSE., so it appears unnecessary to */
+/*       assign it a value when FIRST is .TRUE. */
+/*       Local state */
+/* FIRST  - .TRUE. only on first call to PITSYN. */
+/* IVOICO - Previous VOICE(2) value. */
+/* IPITO  - Previous PITCH value. */
+/* RMSO   - Previous RMS value. */
+/* RCO    - Previous RC values. */
+
+/* JSAMP  - If this routine is called N times with identical values of */
+/*          LFRAME, then the total length of all pitch periods returned */
+/*          is always N*LFRAME-JSAMP, and JSAMP is always in the range 0 
+*/
+/*          to MAXPIT-1 (see below for why this is so).  Thus JSAMP is */
+/*          the number of samples "left over" from the previous call to */
+/*          PITSYN, that haven't been "used" in a pitch period returned */
+/*          from this subroutine.  Every time this subroutine is called, 
+*/
+/*          it returns pitch periods with a total length of at most */
+/*          LFRAME+JSAMP. */
+
+/* IVOICO, IPITO, RCO, and JSAMP need not be assigned an initial value */
+/* with a DATA statement, because they are always initialized on the */
+/* first call to PITSYN. */
+
+/* FIRST and RMSO should be initialized with DATA statements, because */
+/* even on the first call, they are used before being initialized. */
+    /* Parameter adjustments */
+    if (rc) {
+	--rc;
+	}
+    if (rci) {
+	rci_dim1 = *order;
+	rci_offset = rci_dim1 + 1;
+	rci -= rci_offset;
+	}
+    if (voice) {
+	--voice;
+	}
+    if (ivuv) {
+	--ivuv;
+	}
+    if (ipiti) {
+	--ipiti;
+	}
+    if (rmsi) {
+	--rmsi;
+	}
+
+    /* Function Body */
+    ivoico = &(st->ivoico);
+    ipito = &(st->ipito);
+    rmso = &(st->rmso);
+    rco = &(st->rco[0]);
+    jsamp = &(st->jsamp);
+    first = &(st->first_pitsyn);
+
+    if (*rms < 1.f) {
+	*rms = 1.f;
+    }
+    if (*rmso < 1.f) {
+	*rmso = 1.f;
+    }
+    uvpit = 0.f;
+    *ratio = *rms / (*rmso + 8.f);
+    if (*first) {
+	lsamp = 0;
+	ivoice = voice[2];
+	if (ivoice == 0) {
+	    *pitch = *lframe / 4;
+	}
+	*nout = *lframe / *pitch;
+	*jsamp = *lframe - *nout * *pitch;
+
+/*          SYNTHS only calls this subroutine with PITCH in the range 
+20 */
+/*          to 156.  LFRAME = MAXFRM = 180, so NOUT is somewhere in th
+e */
+/*          range 1 to 9. */
+
+/*          JSAMP is "LFRAME mod PITCH", so it is in the range 0 to */
+/*          (PITCH-1), or 0 to MAXPIT-1=155, after the first call. */
+
+	i__1 = *nout;
+	for (i__ = 1; i__ <= i__1; ++i__) {
+	    i__2 = *order;
+	    for (j = 1; j <= i__2; ++j) {
+		rci[j + i__ * rci_dim1] = rc[j];
+	    }
+	    ivuv[i__] = ivoice;
+	    ipiti[i__] = *pitch;
+	    rmsi[i__] = *rms;
+	}
+	*first = FALSE_;
+    } else {
+	vflag = 0;
+	lsamp = *lframe + *jsamp;
+	slope = (*pitch - *ipito) / (real) lsamp;
+	*nout = 0;
+	jused = 0;
+	istart = 1;
+	if (voice[1] == *ivoico && voice[2] == voice[1]) {
+	    if (voice[2] == 0) {
+/* SSUV - -   0  ,  0  ,  0 */
+		*pitch = *lframe / 4;
+		*ipito = *pitch;
+		if (*ratio > 8.f) {
+		    *rmso = *rms;
+		}
+	    }
+/* SSVC - -   1  ,  1  ,  1 */
+	    slope = (*pitch - *ipito) / (real) lsamp;
+	    ivoice = voice[2];
+	} else {
+	    if (*ivoico != 1) {
+		if (*ivoico == voice[1]) {
+/* UV2VC2 - -  0  ,  0  ,  1 */
+		    nl = lsamp - *lframe / 4;
+		} else {
+/* UV2VC1 - -  0  ,  1  ,  1 */
+		    nl = lsamp - *lframe * 3 / 4;
+		}
+		ipiti[1] = nl / 2;
+		ipiti[2] = nl - ipiti[1];
+		ivuv[1] = 0;
+		ivuv[2] = 0;
+		rmsi[1] = *rmso;
+		rmsi[2] = *rmso;
+		i__1 = *order;
+		for (i__ = 1; i__ <= i__1; ++i__) {
+		    rci[i__ + rci_dim1] = rco[i__ - 1];
+		    rci[i__ + (rci_dim1 << 1)] = rco[i__ - 1];
+		    rco[i__ - 1] = rc[i__];
+		}
+		slope = 0.f;
+		*nout = 2;
+		*ipito = *pitch;
+		jused = nl;
+		istart = nl + 1;
+		ivoice = 1;
+	    } else {
+		if (*ivoico != voice[1]) {
+/* VC2UV1 - -   1  ,  0  ,  0 */
+		    lsamp = *lframe / 4 + *jsamp;
+		} else {
+/* VC2UV2 - -   1  ,  1  ,  0 */
+		    lsamp = *lframe * 3 / 4 + *jsamp;
+		}
+		i__1 = *order;
+		for (i__ = 1; i__ <= i__1; ++i__) {
+		    yarc[i__ - 1] = rc[i__];
+		    rc[i__] = rco[i__ - 1];
+		}
+		ivoice = 1;
+		slope = 0.f;
+		vflag = 1;
+	    }
+	}
+/* Here is the value of most variables that are used below, depending 
+on */
+/* the values of IVOICO, VOICE(1), and VOICE(2).  VOICE(1) and VOICE(2
+) */
+/* are input arguments, and IVOICO is the value of VOICE(2) on the */
+/* previous call (see notes for the IF (NOUT .NE. 0) statement near th
+e */
+/* end).  Each of these three values is either 0 or 1.  These three */
+/* values below are given as 3-bit long strings, in the order IVOICO, 
+*/
+/* VOICE(1), and VOICE(2).  It appears that the code above assumes tha
+t */
+/* the bit sequences 010 and 101 never occur, but I wonder whether a 
+*/
+/* large enough number of bit errors in the channel could cause such a
+ */
+/* thing to happen, and if so, could that cause NOUT to ever go over 1
+1? */
+
+/* Note that all of the 180 values in the table are really LFRAME, but
+ */
+/* 180 has fewer characters, and it makes the table a little more */
+/* concrete.  If LFRAME is ever changed, keep this in mind.  Similarly
+, */
+/* 135's are 3*LFRAME/4, and 45's are LFRAME/4.  If LFRAME is not a */
+/* multiple of 4, then the 135 for NL-JSAMP is actually LFRAME-LFRAME/
+4, */
+/* and the 45 for NL-JSAMP is actually LFRAME-3*LFRAME/4. */
+
+/* Note that LSAMP-JSAMP is given as the variable.  This was just for 
+*/
+/* brevity, to avoid adding "+JSAMP" to all of the column entries. */
+/* Similarly for NL-JSAMP. */
+
+/* Variable    | 000  001    011,010  111       110       100,101 */
+/* ------------+-------------------------------------------------- */
+/* ISTART      | 1    NL+1   NL+1     1         1         1 */
+/* LSAMP-JSAMP | 180  180    180      180       135       45 */
+/* IPITO       | 45   PITCH  PITCH    oldPITCH  oldPITCH  oldPITCH */
+/* SLOPE       | 0    0      0        seebelow  0         0 */
+/* JUSED       | 0    NL     NL       0         0         0 */
+/* PITCH       | 45   PITCH  PITCH    PITCH     PITCH     PITCH */
+/* NL-JSAMP    | --   135    45       --        --        -- */
+/* VFLAG       | 0    0      0        0         1         1 */
+/* NOUT        | 0    2      2        0         0         0 */
+/* IVOICE      | 0    1      1        1         1         1 */
+
+/* while_loop  | once once   once     once      twice     twice */
+
+/* ISTART      | --   --     --       --        JUSED+1   JUSED+1 */
+/* LSAMP-JSAMP | --   --     --       --        180       180 */
+/* IPITO       | --   --     --       --        oldPITCH  oldPITCH */
+/* SLOPE       | --   --     --       --        0         0 */
+/* JUSED       | --   --     --       --        ??        ?? */
+/* PITCH       | --   --     --       --        PITCH     PITCH */
+/* NL-JSAMP    | --   --     --       --        --        -- */
+/* VFLAG       | --   --     --       --        0         0 */
+/* NOUT        | --   --     --       --        ??        ?? */
+/* IVOICE      | --   --     --       --        0         0 */
+
+
+/* UVPIT is always 0.0 on the first pass through the DO WHILE (.TRUE.)
+ */
+/* loop below. */
+
+/* The only possible non-0 value of SLOPE (in column 111) is */
+/* (PITCH-IPITO)/FLOAT(LSAMP) */
+
+/* Column 101 is identical to 100.  Any good properties we can prove 
+*/
+/* for 100 will also hold for 101.  Similarly for 010 and 011. */
+
+/* SYNTHS calls this subroutine with PITCH restricted to the range 20 
+to */
+/* 156.  IPITO is similarly restricted to this range, after the first 
+*/
+/* call.  IP below is also restricted to this range, given the */
+/* definitions of IPITO, SLOPE, UVPIT, and that I is in the range ISTA
+RT */
+/* to LSAMP. */
+
+	while(TRUE_) {
+
+/*             JUSED is the total length of all pitch periods curr
+ently */
+/*             in the output arrays, in samples. */
+
+/*             An invariant of the DO I = ISTART,LSAMP loop below,
+ under */
+/*             the condition that IP is always in the range 1 thro
+ugh */
+/*             MAXPIT, is: */
+
+/*             (I - MAXPIT) .LE. JUSED .LE. (I-1) */
+
+/*             Note that the final value of I is LSAMP+1, so that 
+after */
+/*             the DO loop is complete, we know: */
+
+/*             (LSAMP - MAXPIT + 1) .LE. JUSED .LE. LSAMP */
+
+	    i__1 = lsamp;
+	    for (i__ = istart; i__ <= i__1; ++i__) {
+		r__1 = *ipito + slope * i__;
+		ip = r__1 + .5f;
+		if (uvpit != 0.f) {
+		    ip = uvpit;
+		}
+		if (ip <= i__ - jused) {
+		    ++(*nout);
+
+/*                   The following check is no longer nece
+ssary, now that */
+/*                   we can prove that NOUT will never go 
+over 16. */
+
+/* 		    IF (NOUT .GT. 16) STOP 'PITSYN: too many epochs' 
+*/
+
+		    ipiti[*nout] = ip;
+		    *pitch = ip;
+		    ivuv[*nout] = ivoice;
+		    jused += ip;
+		    prop = (jused - ip / 2) / (real) lsamp;
+		    i__2 = *order;
+		    for (j = 1; j <= i__2; ++j) {
+			alro = log((rco[j - 1] + 1) / (1 - rco[j - 1]));
+			alrn = log((rc[j] + 1) / (1 - rc[j]));
+			xxy = alro + prop * (alrn - alro);
+			xxy = exp(xxy);
+			rci[j + *nout * rci_dim1] = (xxy - 1) / (xxy + 1);
+		    }
+		    rmsi[*nout] = log(*rmso) + prop * (log(*rms) - log(*rmso));
+		    rmsi[*nout] = exp(rmsi[*nout]);
+		}
+	    }
+	    if (vflag != 1) {
+		goto L100;
+	    }
+
+/*             I want to prove what range UVPIT must lie in after 
+the */
+/*             assignments to it below.  To do this, I must determ
+ine */
+/*             what range (LSAMP-ISTART) must lie in, after the */
+/*             assignments to ISTART and LSAMP below. */
+
+/*             Let oldLSAMP be the value of LSAMP at this point in
+ the */
+/*             execution.  This is 135+JSAMP in state 110, or 45+J
+SAMP in */
+/*             states 100 or 101. */
+
+/*             Given the loop invariant on JUSED above, we know th
+at: */
+
+/*             (oldLSAMP - MAXPIT + 1) .LE. JUSED .LE. oldLSAMP */
+
+/*             ISTART is one more than this. */
+
+/*             Let newLSAMP be the value assigned to LSAMP below. 
+ This */
+/*             is 180+JSAMP.  Thus (newLSAMP-oldLSAMP) is either 4
+5 or */
+/*             135, depending on the state. */
+
+/*             Thus, the range of newLSAMP-ISTART is: */
+
+/*             (newLSAMP-(oldLSAMP+1)) .LE. newLSAMP-ISTART */
+/*             .LE. (newLSAMP-(oldLSAMP - MAXPIT + 2)) */
+
+/*             or: */
+
+/*             46 .LE. newLSAMP-ISTART .LE. 133+MAXPIT .EQ. 289 */
+
+/*             Therefore, UVPIT is in the range 23 to 144 after th
+e first */
+/*             assignment to UVPIT below, and after the conditiona
+l */
+/*             assignment, it is in the range 23 to 90. */
+
+/*             The important thing is that it is in the range 20 t
+o 156, */
+/*             so that in the loop above, IP is always in this ran
+ge. */
+
+	    vflag = 0;
+	    istart = jused + 1;
+	    lsamp = *lframe + *jsamp;
+	    slope = 0.f;
+	    ivoice = 0;
+	    uvpit = (real) ((lsamp - istart) / 2);
+	    if (uvpit > 90.f) {
+		uvpit /= 2;
+	    }
+	    *rmso = *rms;
+	    i__1 = *order;
+	    for (i__ = 1; i__ <= i__1; ++i__) {
+		rc[i__] = yarc[i__ - 1];
+		rco[i__ - 1] = yarc[i__ - 1];
+	    }
+	}
+L100:
+	*jsamp = lsamp - jused;
+    }
+/*       Given that the maximum pitch period MAXPIT .LT. LFRAME (this is 
+*/
+/*       currently true on every call, since SYNTHS always sets */
+/*       LFRAME=180), NOUT will always be .GE. 1 at this point. */
+    if (*nout != 0) {
+	*ivoico = voice[2];
+	*ipito = *pitch;
+	*rmso = *rms;
+	i__1 = *order;
+	for (i__ = 1; i__ <= i__1; ++i__) {
+	    rco[i__ - 1] = rc[i__];
+	}
+    }
+    return 0;
+} /* pitsyn_ */

codecs/lpc10/placea.c

@@ -0,0 +1,218 @@
+/*
+
+$Log$
+Revision 1.4  2000/01/05 08:20:39  markster
+Version 0.1.5 from FTP
+
+Revision 1.2  2000/01/05 08:20:39  markster
+Some OSS fixes and a few lpc changes to make it actually work
+
+ * Revision 1.1  1996/08/19  22:31:07  jaf
+ * Initial revision
+ *
+
+*/
+
+#ifdef P_R_O_T_O_T_Y_P_E_S
+extern int placea_(integer *ipitch, integer *voibuf, integer *obound, integer *af, integer *vwin, integer *awin, integer *ewin, integer *lframe, integer *maxwin);
+#endif
+
+/*  -- translated by f2c (version 19951025).
+   You must link the resulting object file with the libraries:
+	-lf2c -lm   (in that order)
+*/
+
+#include "f2c.h"
+
+/* *********************************************************************** */
+
+/* 	PLACEA Version 48 */
+
+/* $Log$
+ * Revision 1.4  2000/01/05 08:20:39  markster
+ * Version 0.1.5 from FTP
+ *
+/* Revision 1.2  2000/01/05 08:20:39  markster
+/* Some OSS fixes and a few lpc changes to make it actually work
+/*
+ * Revision 1.1  1996/08/19  22:31:07  jaf
+ * Initial revision
+ * */
+/* Revision 1.5  1996/03/19  20:41:55  jaf */
+/* Added some conditions satisfied by the output values in EWIN. */
+
+/* Revision 1.4  1996/03/19  20:24:17  jaf */
+/* Added some conditions satisfied by the output values in AWIN. */
+
+/* Revision 1.3  1996/03/18  21:40:04  jaf */
+/* Just added a few comments about which array indices of the arguments */
+/* are used, and mentioning that this subroutine has no local state. */
+
+/* Revision 1.2  1996/03/13  16:43:09  jaf */
+/* Comments added explaining that none of the local variables of this */
+/* subroutine need to be saved from one invocation to the next. */
+
+/* Revision 1.1  1996/02/07 14:48:31  jaf */
+/* Initial revision */
+
+
+/* *********************************************************************** */
+/* Input: */
+/*  IPITCH */
+/*  VOIBUF */
+/*          Indices (2,AF-2), (1,AF-1), (2,AF-1), (1,AF), and (2,AF) read.*/
+/*           All other indices untouched. */
+/*  OBOUND */
+/*  AF */
+/*  VWIN */
+/*           Indices (1,AF) and (2,AF) read. */
+/*           All other indices untouched. */
+/*  LFRAME */
+/*  MAXWIN */
+/* Input/Output: */
+/*  AWIN */
+/*           Index (1,AF-1) read. */
+/*           Indices (1,AF) and (2,AF) written, and then read. */
+/*           All other indices untouched. */
+/*           In all cases (except possibly one), the final values will */
+/*           satisfy the condition: AWIN(2,AF)-AWIN(1,AF)+1 = MAXWIN. */
+/*           In that other case, */
+/*           AWIN(1,AF)=VWIN(1,AF) and AWIN(2,AF)=VWIN(2,AF). */
+/* Output: */
+/*  EWIN */
+/*           Indices (1,AF) and (2,AF) written. */
+/*           All other indices untouched. */
+/*           In all cases, the final values will satisfy the condition: */
+/*           AWIN(1,AF) .LE. EWIN(1,AF) .LE. EWIN(2,AF) .LE. AWIN(2,AF) */
+/*           In other words, the energy window is a sub-window of */
+/*           the analysis window. */
+
+/* This subroutine has no local state. */
+
+/* Subroutine */ int placea_(integer *ipitch, integer *voibuf, integer *
+	obound, integer *af, integer *vwin, integer *awin, integer *ewin, 
+	integer *lframe, integer *maxwin)
+{
+    /* System generated locals */
+    real r__1;
+
+    /* Builtin functions */
+    integer i_nint(real *);
+
+    /* Local variables */
+    logical allv, winv;
+    integer i__, j, k, l, hrange;
+    logical ephase;
+    integer lrange;
+
+/*       Arguments */
+/*       Local variables that need not be saved */
+    /* Parameter adjustments */
+    ewin -= 3;
+    awin -= 3;
+    vwin -= 3;
+    --voibuf;
+
+    /* Function Body */
+    lrange = (*af - 2) * *lframe + 1;
+    hrange = *af * *lframe;
+/*   Place the Analysis window based on the voicing window */
+/*   placement, onsets, tentative voicing decision, and pitch. */
+
+/*   Case 1:  Sustained Voiced Speech */
+/*   If the five most recent voicing decisions are */
+/*   voiced, then the window is placed phase-synchronously with the */
+/*   previous window, as close to the present voicing window if possible. 
+*/
+/*   If onsets bound the voicing window, then preference is given to */
+/*   a phase-synchronous placement which does not overlap these onsets. */
+
+/*   Case 2:  Voiced Transition */
+/*   If at least one voicing decision in AF is voicied, and there are no 
+*/
+/*   onsets, then the window is placed as in case 1. */
+
+/*   Case 3:  Unvoiced Speech or Onsets */
+/*   If both voicing decisions in AF are unvoiced, or there are onsets, */
+/*   then the window is placed coincident with the voicing window. */
+
+/*   Note:  During phase-synchronous placement of windows, the length */
+/*   is not altered from MAXWIN, since this would defeat the purpose */
+/*   of phase-synchronous placement. */
+/* Check for case 1 and case 2 */
+    allv = voibuf[(*af - 2 << 1) + 2] == 1;
+    allv = allv && voibuf[(*af - 1 << 1) + 1] == 1;
+    allv = allv && voibuf[(*af - 1 << 1) + 2] == 1;
+    allv = allv && voibuf[(*af << 1) + 1] == 1;
+    allv = allv && voibuf[(*af << 1) + 2] == 1;
+    winv = voibuf[(*af << 1) + 1] == 1 || voibuf[(*af << 1) + 2] == 1;
+    if (allv || winv && *obound == 0) {
+/* APHASE:  Phase synchronous window placement. */
+/* Get minimum lower index of the window. */
+	i__ = (lrange + *ipitch - 1 - awin[(*af - 1 << 1) + 1]) / *ipitch;
+	i__ *= *ipitch;
+	i__ += awin[(*af - 1 << 1) + 1];
+/* L = the actual length of this frame's analysis window. */
+	l = *maxwin;
+/* Calculate the location where a perfectly centered window would star
+t. */
+	k = (vwin[(*af << 1) + 1] + vwin[(*af << 1) + 2] + 1 - l) / 2;
+/* Choose the actual location to be the pitch multiple closest to this
+. */
+	r__1 = (real) (k - i__) / *ipitch;
+	awin[(*af << 1) + 1] = i__ + i_nint(&r__1) * *ipitch;
+	awin[(*af << 1) + 2] = awin[(*af << 1) + 1] + l - 1;
+/* If there is an onset bounding the right of the voicing window and t
+he */
+/* analysis window overlaps that, then move the analysis window backwa
+rd */
+/* to avoid this onset. */
+	if (*obound >= 2 && awin[(*af << 1) + 2] > vwin[(*af << 1) + 2]) {
+	    awin[(*af << 1) + 1] -= *ipitch;
+	    awin[(*af << 1) + 2] -= *ipitch;
+	}
+/* Similarly for the left of the voicing window. */
+	if ((*obound == 1 || *obound == 3) && awin[(*af << 1) + 1] < vwin[(*
+		af << 1) + 1]) {
+	    awin[(*af << 1) + 1] += *ipitch;
+	    awin[(*af << 1) + 2] += *ipitch;
+	}
+/* If this placement puts the analysis window above HRANGE, then */
+/* move it backward an integer number of pitch periods. */
+	while(awin[(*af << 1) + 2] > hrange) {
+	    awin[(*af << 1) + 1] -= *ipitch;
+	    awin[(*af << 1) + 2] -= *ipitch;
+	}
+/* Similarly if the placement puts the analysis window below LRANGE. 
+*/
+	while(awin[(*af << 1) + 1] < lrange) {
+	    awin[(*af << 1) + 1] += *ipitch;
+	    awin[(*af << 1) + 2] += *ipitch;
+	}
+/* Make Energy window be phase-synchronous. */
+	ephase = TRUE_;
+/* Case 3 */
+    } else {
+	awin[(*af << 1) + 1] = vwin[(*af << 1) + 1];
+	awin[(*af << 1) + 2] = vwin[(*af << 1) + 2];
+	ephase = FALSE_;
+    }
+/* RMS is computed over an integer number of pitch periods in the analysis
+ */
+/*window.  When it is not placed phase-synchronously, it is placed as clos
+e*/
+/* as possible to onsets. */
+    j = (awin[(*af << 1) + 2] - awin[(*af << 1) + 1] + 1) / *ipitch * *ipitch;
+    if (j == 0 || ! winv) {
+	ewin[(*af << 1) + 1] = vwin[(*af << 1) + 1];
+	ewin[(*af << 1) + 2] = vwin[(*af << 1) + 2];
+    } else if (! ephase && *obound == 2) {
+	ewin[(*af << 1) + 1] = awin[(*af << 1) + 2] - j + 1;
+	ewin[(*af << 1) + 2] = awin[(*af << 1) + 2];
+    } else {
+	ewin[(*af << 1) + 1] = awin[(*af << 1) + 1];
+	ewin[(*af << 1) + 2] = awin[(*af << 1) + 1] + j - 1;
+    }
+    return 0;
+} /* placea_ */
+

codecs/lpc10/placev.c

@@ -0,0 +1,257 @@
+/*
+
+$Log$
+Revision 1.4  2000/01/05 08:20:39  markster
+Version 0.1.5 from FTP
+
+Revision 1.2  2000/01/05 08:20:39  markster
+Some OSS fixes and a few lpc changes to make it actually work
+
+ * Revision 1.1  1996/08/19  22:31:02  jaf
+ * Initial revision
+ *
+
+*/
+
+#ifdef P_R_O_T_O_T_Y_P_E_S
+extern int placev_(integer *osbuf, integer *osptr, integer *oslen, integer *obound, integer *vwin, integer *af, integer *lframe, integer *minwin, integer *maxwin, integer *dvwinl, integer *dvwinh);
+#endif
+
+/*  -- translated by f2c (version 19951025).
+   You must link the resulting object file with the libraries:
+	-lf2c -lm   (in that order)
+*/
+
+#include "f2c.h"
+
+/* ****************************************************************** */
+
+/* 	PLACEV Version 48 */
+
+/* $Log$
+ * Revision 1.4  2000/01/05 08:20:39  markster
+ * Version 0.1.5 from FTP
+ *
+/* Revision 1.2  2000/01/05 08:20:39  markster
+/* Some OSS fixes and a few lpc changes to make it actually work
+/*
+ * Revision 1.1  1996/08/19  22:31:02  jaf
+ * Initial revision
+ * */
+/* Revision 1.6  1996/03/19  20:42:19  jaf */
+/* Added some conditions satisfied by the output values in VWIN. */
+
+/* Revision 1.5  1996/03/19  18:37:56  jaf */
+/* Strengthened the specification of which indices of VWIN are read and */
+/* written. */
+
+/* Revision 1.4  1996/03/15  16:38:33  jaf */
+/* One tiny comment added. */
+
+/* Revision 1.3  1996/03/15  16:36:13  jaf */
+/* Added comments giving In/Out status of arguments. */
+
+/* Revision 1.2  1996/03/12  23:56:01  jaf */
+/* Comments added explaining that none of the local variables of this */
+/* subroutine need to be saved from one invocation to the next. */
+
+/* Revision 1.1  1996/02/07 14:48:39  jaf */
+/* Initial revision */
+
+
+/* ****************************************************************** */
+
+/* Input: */
+/*  OSBUF	Buffer which holds sorted indexes of onsets */
+/*       	I believe that only indices 1 through OSPTR-1 can be read. */
+/*  OSLEN */
+/*  OSPTR	Free pointer into OSBUF */
+/*  AF */
+/*  LFRAME */
+/*  MINWIN */
+/*  MAXWIN */
+/*  DVWINL */
+/*  DVWINH	(This argument is never used.  Should it be?) */
+/* Input/Output: */
+/*  VWIN		Buffer of Voicing Window Positions (Modified) */
+/*       	Index (2,AF-1) is read. */
+/*       	Indices (1,AF) and (2,AF) are written, */
+/*       	and then possibly read. */
+/*       	All other indices are unused. */
+/*              In all cases, the final values will satsify the condition:*/
+/*               VWIN(2,AF)-VWIN(1,AF)+1 .LE. MAXWIN */
+/*               I'm not certain yet, but they may also satisfy: */
+/*               MINWIN .LE. VWIN(2,AF)-VWIN(1,AF)+1 */
+/* Output: */
+/*  OBOUND	This variable is set by this procedure and used */
+/* 		in placing analysis windows (PLACEA).  Bit 1 */
+/* 		indicates whether an onset bounds the left side */
+/* 		of the voicing window, and bit 2 indicates whether */
+/* 		an onset bounds the right side of the voicing window. */
+
+/* This subroutine has no local state. */
+
+/* Subroutine */ int placev_(integer *osbuf, integer *osptr, integer *oslen, 
+	integer *obound, integer *vwin, integer *af, integer *lframe, integer 
+	*minwin, integer *maxwin, integer *dvwinl, integer *dvwinh)
+{
+    /* System generated locals */
+    integer i__1, i__2;
+
+    /* Local variables */
+    logical crit;
+    integer i__, q, osptr1, hrange, lrange;
+
+/*       Arguments */
+/*       Local variables that need not be saved */
+/*   Variables */
+/*    LRANGE, HRANGE  Range in which window is placed */
+/*    OSPTR1     OSPTR excluding samples in 3F */
+/*       Local state */
+/*       None */
+/*   Voicing Window Placement */
+
+/*         __________________ __________________ ______________ */
+/*        |                  |                  | */
+/*        |        1F        |        2F        |        3F ... */
+/*        |__________________|__________________|______________ */
+
+/*    Previous | */
+/*      Window | */
+/*  ...________| */
+
+/*             |                                | */
+/*      ------>| This window's placement range  |<------ */
+/*             |                                | */
+
+/*   There are three cases.  Note that these are different from those */
+/*   given in the LPC-10e phase 1 report. */
+
+/*   1.  If there are no onsets in this range, then the voicing window */
+/*   is centered in the pitch window.  If such a placement is not within 
+*/
+/*   the window's placement range, then the window is placed in the left- 
+*/
+/*   most portion of the placement range.  Its length is always MAXWIN. */
+
+/*   2.  If the first onset is in 2F and there is sufficient room to place
+ */
+/*   the window immediately before this onset, then the window is placed 
+*/
+/*   there, and its length is set to the maximum possible under these */
+/*   constraints. */
+
+/* 	"Critical Region Exception":  If there is another onset in 2F */
+/* 	such that a window can be placed between the two onsets, the */
+/* 	window is placed there (ie, as in case 3). */
+
+/*   3.  Otherwise, the window is placed immediately after the onset.  The
+ */
+/*   window's length */
+/*  is the longest length that can fit in the range under these constraint
+s,*/
+/*  except that the window may be shortened even further to avoid overlapp
+ing*/
+/*  other onsets in the placement range.  In any case, the window's length
+*/
+/*   is at least MINWIN. */
+
+/*   Note that the values of MINWIN and LFRAME must be chosen such */
+/*   that case 2 = false implies case 3 = true.   This means that */
+/*   MINWIN <= LFRAME/2.  If this were not the case, then a fourth case */
+/*   would have to be added for when the window cannot fit either before 
+*/
+/*   or after the onset. */
+
+/*   Note also that onsets which weren't in 2F last time may be in 1F this
+ */
+/*  time, due to the filter delays in computing onsets.  The result is tha
+t*/
+/*   occasionally a voicing window will overlap that onset.  The only way 
+*/
+/*   to circumvent this problem is to add more delay in processing input 
+*/
+/*   speech.  In the trade-off between delay and window-placement, window 
+*/
+/*   placement lost. */
+/* Compute the placement range */
+    /* Parameter adjustments */
+    --osbuf;
+    vwin -= 3;
+
+    /* Function Body */
+/* Computing MAX */
+    i__1 = vwin[(*af - 1 << 1) + 2] + 1, i__2 = (*af - 2) * *lframe + 1;
+    lrange = max(i__1,i__2);
+    hrange = *af * *lframe;
+/* Compute OSPTR1, so the following code only looks at relevant onsets. */
+    for (osptr1 = *osptr - 1; osptr1 >= 1; --osptr1) {
+	if (osbuf[osptr1] <= hrange) {
+	    goto L90;
+	}
+    }
+L90:
+    ++osptr1;
+/* Check for case 1 first (fast case): */
+    if (osptr1 <= 1 || osbuf[osptr1 - 1] < lrange) {
+/* Computing MAX */
+	i__1 = vwin[(*af - 1 << 1) + 2] + 1;
+	vwin[(*af << 1) + 1] = max(i__1,*dvwinl);
+	vwin[(*af << 1) + 2] = vwin[(*af << 1) + 1] + *maxwin - 1;
+	*obound = 0;
+    } else {
+/* Search backward in OSBUF for first onset in range. */
+/* This code relies on the above check being performed first. */
+	for (q = osptr1 - 1; q >= 1; --q) {
+	    if (osbuf[q] < lrange) {
+		goto L100;
+	    }
+	}
+L100:
+	++q;
+/* Check for case 2 (placement before onset): */
+/* Check for critical region exception: */
+	i__1 = osptr1 - 1;
+	for (i__ = q + 1; i__ <= i__1; ++i__) {
+	    if (osbuf[i__] - osbuf[q] >= *minwin) {
+		crit = TRUE_;
+		goto L105;
+	    }
+	}
+	crit = FALSE_;
+L105:
+/* Computing MAX */
+	i__1 = (*af - 1) * *lframe, i__2 = lrange + *minwin - 1;
+	if (! crit && osbuf[q] > max(i__1,i__2)) {
+	    vwin[(*af << 1) + 2] = osbuf[q] - 1;
+/* Computing MAX */
+	    i__1 = lrange, i__2 = vwin[(*af << 1) + 2] - *maxwin + 1;
+	    vwin[(*af << 1) + 1] = max(i__1,i__2);
+	    *obound = 2;
+/* Case 3 (placement after onset) */
+	} else {
+	    vwin[(*af << 1) + 1] = osbuf[q];
+L110:
+	    ++q;
+	    if (q >= osptr1) {
+		goto L120;
+	    }
+	    if (osbuf[q] > vwin[(*af << 1) + 1] + *maxwin) {
+		goto L120;
+	    }
+	    if (osbuf[q] < vwin[(*af << 1) + 1] + *minwin) {
+		goto L110;
+	    }
+	    vwin[(*af << 1) + 2] = osbuf[q] - 1;
+	    *obound = 3;
+	    return 0;
+L120:
+/* Computing MIN */
+	    i__1 = vwin[(*af << 1) + 1] + *maxwin - 1;
+	    vwin[(*af << 1) + 2] = min(i__1,hrange);
+	    *obound = 1;
+	}
+    }
+    return 0;
+} /* placev_ */
+

codecs/lpc10/preemp.c

@@ -0,0 +1,132 @@
+/*
+
+$Log$
+Revision 1.4  2000/01/05 08:20:39  markster
+Version 0.1.5 from FTP
+
+Revision 1.2  2000/01/05 08:20:39  markster
+Some OSS fixes and a few lpc changes to make it actually work
+
+ * Revision 1.1  1996/08/19  22:30:58  jaf
+ * Initial revision
+ *
+
+*/
+
+#ifdef P_R_O_T_O_T_Y_P_E_S
+extern int preemp_(real *inbuf, real *pebuf, integer *nsamp, real *coef, real *z__);
+#endif
+
+/*  -- translated by f2c (version 19951025).
+   You must link the resulting object file with the libraries:
+	-lf2c -lm   (in that order)
+*/
+
+#include "f2c.h"
+
+/* ******************************************************************* */
+
+/* 	PREEMP Version 55 */
+
+/* $Log$
+ * Revision 1.4  2000/01/05 08:20:39  markster
+ * Version 0.1.5 from FTP
+ *
+/* Revision 1.2  2000/01/05 08:20:39  markster
+/* Some OSS fixes and a few lpc changes to make it actually work
+/*
+ * Revision 1.1  1996/08/19  22:30:58  jaf
+ * Initial revision
+ * */
+/* Revision 1.3  1996/03/14  23:16:29  jaf */
+/* Just added a few comments about which array indices of the arguments */
+/* are used, and mentioning that this subroutine has no local state. */
+
+/* Revision 1.2  1996/03/11  23:23:34  jaf */
+/* Added a bunch of comments to an otherwise simple subroutine. */
+
+/* Revision 1.1  1996/02/07 14:48:48  jaf */
+/* Initial revision */
+
+
+/* ******************************************************************* */
+
+/*   Preemphasize speech with a single-zero filter. */
+/*  (When coef = .9375, preemphasis is as in LPC43.) */
+
+/* Inputs: */
+/*  NSAMP  - Number of samples to filter */
+/*  INBUF  - Input speech buffer */
+/*           Indices 1 through NSAMP are read. */
+/*  COEF   - Preemphasis coefficient */
+/* Input/Output: */
+/*  Z      - Filter state */
+/* Output: */
+/*  PEBUF  - Preemphasized speech buffer (can be equal to INBUF) */
+/*           Indices 1 through NSAMP are modified. */
+
+/* This subroutine has no local state. */
+
+/* Subroutine */ int preemp_(real *inbuf, real *pebuf, integer *nsamp, real *
+	coef, real *z__)
+{
+    /* System generated locals */
+    integer i__1;
+
+    /* Local variables */
+    real temp;
+    integer i__;
+
+/*       Arguments */
+/*       Local variables */
+
+/*       None of these need to have their values saved from one */
+/*       invocation to the next. */
+
+/*       Logically, this subroutine computes the output sequence */
+/*       pebuf(1:nsamp) defined by: */
+
+/*       pebuf(i) = inbuf(i) - coef * inbuf(i-1) */
+
+/*       where inbuf(0) is defined by the value of z given as input to */
+/*       this subroutine. */
+
+/*       What is this filter's frequency response and phase response? */
+
+/*       Why is this filter applied to the speech? */
+
+/*       Could it be more efficient to apply multiple filters */
+/*       simultaneously, by combining them into one equivalent filter? */
+
+/*       Are there ever cases when "factoring" one high-order filter into 
+*/
+/*       multiple smaller-order filter actually reduces the number of */
+/*       arithmetic operations needed to perform them? */
+/*       When I first read this subroutine, I didn't understand why the */
+/*       variable temp was used.  It seemed that the statements in the do 
+*/
+/*       loop could be replaced with the following: */
+
+/*           pebuf(i) = inbuf(i) - coef * z */
+/*           z = inbuf(i) */
+
+/*       The reason for temp is so that even if pebuf and inbuf are the */
+/*       same arrays in memory (i.e., they are aliased), then this */
+/*       subroutine will still work correctly.  I didn't realize this */
+/*       until seeing the comment after PEBUF above that says "(can be */
+/*       equal to INBUF)". */
+    /* Parameter adjustments */
+    --pebuf;
+    --inbuf;
+
+    /* Function Body */
+    i__1 = *nsamp;
+    for (i__ = 1; i__ <= i__1; ++i__) {
+	temp = inbuf[i__] - *coef * *z__;
+	*z__ = inbuf[i__];
+	pebuf[i__] = temp;
+/* L10: */
+    }
+    return 0;
+} /* preemp_ */
+

codecs/lpc10/prepro.c

@@ -0,0 +1,105 @@
+/*
+
+$Log$
+Revision 1.4  2000/01/05 08:20:39  markster
+Version 0.1.5 from FTP
+
+Revision 1.2  2000/01/05 08:20:39  markster
+Some OSS fixes and a few lpc changes to make it actually work
+
+ * Revision 1.2  1996/08/20  20:40:51  jaf
+ * Removed all static local variables that were SAVE'd in the Fortran
+ * code, and put them in struct lpc10_encoder_state that is passed as an
+ * argument.
+ *
+ * Removed init function, since all initialization is now done in
+ * init_lpc10_encoder_state().
+ *
+ * Revision 1.1  1996/08/19  22:30:54  jaf
+ * Initial revision
+ *
+
+*/
+
+#ifdef P_R_O_T_O_T_Y_P_E_S
+extern int prepro_(real *speech, integer *length,
+		   struct lpc10_encoder_state *st)
+/*:ref: hp100_ 14 3 6 4 4 */
+/*:ref: inithp100_ 14 0 */
+#endif
+
+/*  -- translated by f2c (version 19951025).
+   You must link the resulting object file with the libraries:
+	-lf2c -lm   (in that order)
+*/
+
+#include "f2c.h"
+
+/* Table of constant values */
+
+static integer c__1 = 1;
+
+/* ********************************************************************* */
+
+/* 	PREPRO Version 48 */
+
+/* $Log$
+ * Revision 1.4  2000/01/05 08:20:39  markster
+ * Version 0.1.5 from FTP
+ *
+/* Revision 1.2  2000/01/05 08:20:39  markster
+/* Some OSS fixes and a few lpc changes to make it actually work
+/*
+ * Revision 1.2  1996/08/20  20:40:51  jaf
+ * Removed all static local variables that were SAVE'd in the Fortran
+ * code, and put them in struct lpc10_encoder_state that is passed as an
+ * argument.
+ *
+ * Removed init function, since all initialization is now done in
+ * init_lpc10_encoder_state().
+ *
+ * Revision 1.1  1996/08/19  22:30:54  jaf
+ * Initial revision
+ * */
+/* Revision 1.3  1996/03/14  23:22:56  jaf */
+/* Added comments about when INITPREPRO should be used. */
+
+/* Revision 1.2  1996/03/14  23:09:27  jaf */
+/* Added an entry named INITPREPRO that initializes the local state of */
+/* this subroutine, and those it calls (if any). */
+
+/* Revision 1.1  1996/02/07  14:48:54  jaf */
+/* Initial revision */
+
+
+/* ********************************************************************* */
+
+/*    Pre-process input speech: */
+
+/* Inputs: */
+/*  LENGTH - Number of SPEECH samples */
+/* Input/Output: */
+/*  SPEECH(LENGTH) - Speech data. */
+/*                   Indices 1 through LENGTH are read and modified. */
+
+/* This subroutine has no local state maintained from one call to the */
+/* next, but HP100 does.  If you want to switch to using a new audio */
+/* stream for this filter, or reinitialize its state for any other */
+/* reason, call the ENTRY INITPREPRO. */
+
+/* Subroutine */ int prepro_(real *speech, integer *length,
+			     struct lpc10_encoder_state *st)
+{
+    extern /* Subroutine */ int hp100_(real *, integer *, integer *, struct lpc10_encoder_state *);
+
+/*       Arguments */
+/*   High Pass Filter at 100 Hz */
+    /* Parameter adjustments */
+    if (speech) {
+	--speech;
+	}
+
+    /* Function Body */
+    hp100_(&speech[1], &c__1, length, st);
+    return 0;
+} /* prepro_ */

codecs/lpc10/random.c

@@ -0,0 +1,113 @@
+/*
+
+$Log$
+Revision 1.4  2000/01/05 08:20:39  markster
+Version 0.1.5 from FTP
+
+Revision 1.2  2000/01/05 08:20:39  markster
+Some OSS fixes and a few lpc changes to make it actually work
+
+ * Revision 1.2  1996/08/20  20:41:32  jaf
+ * Removed all static local variables that were SAVE'd in the Fortran
+ * code, and put them in struct lpc10_decoder_state that is passed as an
+ * argument.
+ *
+ * Removed init function, since all initialization is now done in
+ * init_lpc10_decoder_state().
+ *
+ * Revision 1.1  1996/08/19  22:30:49  jaf
+ * Initial revision
+ *
+
+*/
+
+#ifdef P_R_O_T_O_T_Y_P_E_S
+extern integer random_(struct lpc10_decoder_state *st);
+#endif
+
+/*  -- translated by f2c (version 19951025).
+   You must link the resulting object file with the libraries:
+	-lf2c -lm   (in that order)
+*/
+
+#include "f2c.h"
+
+/* ********************************************************************** */
+
+/* 	RANDOM Version 49 */
+
+/* $Log$
+ * Revision 1.4  2000/01/05 08:20:39  markster
+ * Version 0.1.5 from FTP
+ *
+/* Revision 1.2  2000/01/05 08:20:39  markster
+/* Some OSS fixes and a few lpc changes to make it actually work
+/*
+ * Revision 1.2  1996/08/20  20:41:32  jaf
+ * Removed all static local variables that were SAVE'd in the Fortran
+ * code, and put them in struct lpc10_decoder_state that is passed as an
+ * argument.
+ *
+ * Removed init function, since all initialization is now done in
+ * init_lpc10_decoder_state().
+ *
+ * Revision 1.1  1996/08/19  22:30:49  jaf
+ * Initial revision
+ * */
+/* Revision 1.3  1996/03/20  16:13:54  jaf */
+/* Rearranged comments a little bit, and added comments explaining that */
+/* even though there is local state here, there is no need to create an */
+/* ENTRY for reinitializing it. */
+
+/* Revision 1.2  1996/03/14  22:25:29  jaf */
+/* Just rearranged the comments and local variable declarations a bit. */
+
+/* Revision 1.1  1996/02/07 14:49:01  jaf */
+/* Initial revision */
+
+
+/* ********************************************************************* */
+
+/*  Pseudo random number generator based on Knuth, Vol 2, p. 27. */
+
+/* Function Return: */
+/*  RANDOM - Integer variable, uniformly distributed over -32768 to 32767 */
+
+/* This subroutine maintains local state from one call to the next. */
+/* In the context of the LPC10 coder, there is no reason to reinitialize */
+/* this local state when switching between audio streams, because its */
+/* results are only used to generate noise for unvoiced frames. */
+
+integer random_(struct lpc10_decoder_state *st)
+{
+    /* Initialized data */
+
+    integer *j;
+    integer *k;
+    shortint *y;
+
+    /* System generated locals */
+    integer ret_val;
+
+/* 	Parameters/constants */
+/*       Local state */
+/*   The following is a 16 bit 2's complement addition, */
+/*   with overflow checking disabled */
+
+    j = &(st->j);
+    k = &(st->k);
+    y = &(st->y[0]);
+
+    y[*k - 1] += y[*j - 1];
+    ret_val = y[*k - 1];
+    --(*k);
+    if (*k <= 0) {
+	*k = 5;
+    }
+    --(*j);
+    if (*j <= 0) {
+	*j = 5;
+    }
+    return ret_val;
+} /* random_ */
+

codecs/lpc10/rcchk.c

@@ -0,0 +1,107 @@
+/*
+
+$Log$
+Revision 1.4  2000/01/05 08:20:39  markster
+Version 0.1.5 from FTP
+
+Revision 1.2  2000/01/05 08:20:39  markster
+Some OSS fixes and a few lpc changes to make it actually work
+
+ * Revision 1.1  1996/08/19  22:30:41  jaf
+ * Initial revision
+ *
+
+*/
+
+#ifdef P_R_O_T_O_T_Y_P_E_S
+extern int rcchk_(integer *order, real *rc1f, real *rc2f);
+#endif
+
+/*  -- translated by f2c (version 19951025).
+   You must link the resulting object file with the libraries:
+	-lf2c -lm   (in that order)
+*/
+
+#include "f2c.h"
+
+/* ********************************************************************* */
+
+/* 	RCCHK Version 45G */
+
+/* $Log$
+ * Revision 1.4  2000/01/05 08:20:39  markster
+ * Version 0.1.5 from FTP
+ *
+/* Revision 1.2  2000/01/05 08:20:39  markster
+/* Some OSS fixes and a few lpc changes to make it actually work
+/*
+ * Revision 1.1  1996/08/19  22:30:41  jaf
+ * Initial revision
+ * */
+/* Revision 1.4  1996/03/27  18:13:47  jaf */
+/* Commented out a call to subroutine ERROR. */
+
+/* Revision 1.3  1996/03/18  15:48:53  jaf */
+/* Just added a few comments about which array indices of the arguments */
+/* are used, and mentioning that this subroutine has no local state. */
+
+/* Revision 1.2  1996/03/13  16:55:22  jaf */
+/* Comments added explaining that none of the local variables of this */
+/* subroutine need to be saved from one invocation to the next. */
+
+/* Revision 1.1  1996/02/07 14:49:08  jaf */
+/* Initial revision */
+
+
+/* ********************************************************************* */
+
+/*  Check RC's, repeat previous frame's RC's if unstable */
+
+/* Input: */
+/*  ORDER - Number of RC's */
+/*  RC1F  - Previous frame's RC's */
+/*          Indices 1 through ORDER may be read. */
+/* Input/Output: */
+/*  RC2F  - Present frame's RC's */
+/*          Indices 1 through ORDER may be read, and written. */
+
+/* This subroutine has no local state. */
+
+/* Subroutine */ int rcchk_(integer *order, real *rc1f, real *rc2f)
+{
+    /* System generated locals */
+    integer i__1;
+    real r__1;
+
+    /* Local variables */
+    integer i__;
+
+/*       Arguments */
+/*       Local variables that need not be saved */
+    /* Parameter adjustments */
+    --rc2f;
+    --rc1f;
+
+    /* Function Body */
+    i__1 = *order;
+    for (i__ = 1; i__ <= i__1; ++i__) {
+	if ((r__1 = rc2f[i__], abs(r__1)) > .99f) {
+	    goto L10;
+	}
+    }
+    return 0;
+/*       Note: In version embedded in other software, all calls to ERROR 
+*/
+/*       should probably be removed. */
+L10:
+
+/*       This call to ERROR is only needed for debugging purposes. */
+
+/*       CALL ERROR('RCCHK',2,I) */
+    i__1 = *order;
+    for (i__ = 1; i__ <= i__1; ++i__) {
+	rc2f[i__] = rc1f[i__];
+    }
+    return 0;
+} /* rcchk_ */
+

codecs/lpc10/synths.c

@@ -0,0 +1,392 @@
+/*
+
+$Log$
+Revision 1.4  2000/01/05 08:20:39  markster
+Version 0.1.5 from FTP
+
+Revision 1.2  2000/01/05 08:20:39  markster
+Some OSS fixes and a few lpc changes to make it actually work
+
+ * Revision 1.2  1996/08/20  20:42:59  jaf
+ * Removed all static local variables that were SAVE'd in the Fortran
+ * code, and put them in struct lpc10_decoder_state that is passed as an
+ * argument.
+ *
+ * Removed init function, since all initialization is now done in
+ * init_lpc10_decoder_state().
+ *
+ * Revision 1.1  1996/08/19  22:30:33  jaf
+ * Initial revision
+ *
+
+*/
+
+#ifdef P_R_O_T_O_T_Y_P_E_S
+extern int synths_(integer *voice, integer *pitch, real *rms, real *rc, real *speech, integer *k, struct lpc10_decoder_state *st);
+/* comlen contrl_ 12 */
+/*:ref: pitsyn_ 14 12 4 4 4 6 6 4 4 4 6 6 4 6 */
+/*:ref: irc2pc_ 14 5 6 6 4 6 6 */
+/*:ref: bsynz_ 14 7 6 4 4 6 6 6 6 */
+/*:ref: deemp_ 14 2 6 4 */
+/*:ref: initpitsyn_ 14 0 */
+/*:ref: initbsynz_ 14 0 */
+/*:ref: initdeemp_ 14 0 */
+#endif
+
+/*  -- translated by f2c (version 19951025).
+   You must link the resulting object file with the libraries:
+	-lf2c -lm   (in that order)
+*/
+
+#include "f2c.h"
+
+/* Common Block Declarations */
+
+extern struct {
+    integer order, lframe;
+    logical corrp;
+} contrl_;
+
+#define contrl_1 contrl_
+
+/* Table of constant values */
+
+static real c_b2 = .7f;
+
+/* ***************************************************************** */
+
+/* 	SYNTHS Version 54 */
+
+/* $Log$
+ * Revision 1.4  2000/01/05 08:20:39  markster
+ * Version 0.1.5 from FTP
+ *
+/* Revision 1.2  2000/01/05 08:20:39  markster
+/* Some OSS fixes and a few lpc changes to make it actually work
+/*
+ * Revision 1.2  1996/08/20  20:42:59  jaf
+ * Removed all static local variables that were SAVE'd in the Fortran
+ * code, and put them in struct lpc10_decoder_state that is passed as an
+ * argument.
+ *
+ * Removed init function, since all initialization is now done in
+ * init_lpc10_decoder_state().
+ *
+ * Revision 1.1  1996/08/19  22:30:33  jaf
+ * Initial revision
+ * */
+/* Revision 1.5  1996/03/26  19:31:58  jaf */
+/* Commented out trace statements. */
+
+/* Revision 1.4  1996/03/25  19:41:01  jaf */
+/* Changed so that MAXFRM samples are always returned in the output array */
+/* SPEECH. */
+
+/* This required delaying the returned samples by MAXFRM sample times, */
+/* and remembering any "left over" samples returned by PITSYN from one */
+/* call of SYNTHS to the next. */
+
+/* Changed size of SPEECH from 2*MAXFRM to MAXFRM.  Removed local */
+/* variable SOUT.  Added local state variables BUF and BUFLEN. */
+
+/* Revision 1.3  1996/03/25  19:20:10  jaf */
+/* Added comments about the range of possible return values for argument */
+/* K, and increased the size of the arrays filled in by PITSYN from 11 to */
+/* 16, as has been already done inside of PITSYN. */
+
+/* Revision 1.2  1996/03/22  00:18:18  jaf */
+/* Added comments explaining meanings of input and output parameters, and */
+/* indicating which array indices can be read or written. */
+
+/* Added entry INITSYNTHS, which does nothing except call the */
+/* corresponding initialization entries for subroutines PITSYN, BSYNZ, */
+/* and DEEMP. */
+
+/* Revision 1.1  1996/02/07 14:49:44  jaf */
+/* Initial revision */
+
+
+/* ***************************************************************** */
+
+/* The note below is from the distributed version of the LPC10 coder. */
+/* The version of the code below has been modified so that SYNTHS always */
+/* has a constant frame length output of MAXFRM. */
+
+/* Also, BSYNZ and DEEMP need not be modified to work on variable */
+/* positions within an array.  It is only necessary to pass the first */
+/* index desired as the array argument.  What actually gets passed is the */
+/* address of that array position, which the subroutine treats as the */
+/* first index of the array. */
+
+/* This technique is used in subroutine ANALYS when calling PREEMP, so it */
+/* appears that multiple people wrote different parts of this LPC10 code, */
+/* and that they didn't necessarily have equivalent knowledge of Fortran */
+/* (not surprising). */
+
+/*  NOTE: There is excessive buffering here, BSYNZ and DEEMP should be */
+/*        changed to operate on variable positions within SOUT.  Also, */
+/*        the output length parameter is bogus, and PITSYN should be */
+/*        rewritten to allow a constant frame length output. */
+
+/* Input: */
+/*  VOICE  - Half frame voicing decisions */
+/*           Indices 1 through 2 read. */
+/* Input/Output: */
+/*  PITCH  - Pitch */
+/*           PITCH is restricted to range 20 to 156, inclusive, */
+/*           before calling subroutine PITSYN, and then PITSYN */
+/*           can modify it further under some conditions. */
+/*  RMS    - Energy */
+/*           Only use is for debugging, and passed to PITSYN. */
+/*           See comments there for how it can be modified. */
+/*  RC     - Reflection coefficients */
+/*           Indices 1 through ORDER restricted to range -.99 to .99, */
+/*           before calling subroutine PITSYN, and then PITSYN */
+/*           can modify it further under some conditions. */
+/* Output: */
+/*  SPEECH - Synthesized speech samples. */
+/*           Indices 1 through the final value of K are written. */
+/*  K      - Number of samples placed into array SPEECH. */
+/*           This is always MAXFRM. */
+
+/* Subroutine */ int synths_(integer *voice, integer *pitch, real *
+	rms, real *rc, real *speech, integer *k, struct lpc10_decoder_state *st)
+{
+    /* Initialized data */
+
+    real *buf;
+    integer *buflen;
+
+    /* System generated locals */
+    integer i__1;
+    real r__1, r__2;
+
+    /* Local variables */
+    real rmsi[16];
+    integer nout, ivuv[16], i__, j;
+    extern /* Subroutine */ int deemp_(real *, integer *, struct lpc10_decoder_state *);
+    real ratio;
+    integer ipiti[16];
+    extern /* Subroutine */ bsynz_(real *, integer *, 
+	    integer *, real *, real *, real *, real *, struct lpc10_decoder_state *), irc2pc_(real *, real *
+	    , integer *, real *, real *);
+    real g2pass;
+    real pc[10];
+    extern /* Subroutine */ int pitsyn_(integer *, integer *, integer *, real 
+	    *, real *, integer *, integer *, integer *, real *, real *, 
+	    integer *, real *, struct lpc10_decoder_state *);
+    real rci[160]	/* was [10][16] */;
+
+/* $Log$
+ * Revision 1.4  2000/01/05 08:20:39  markster
+ * Version 0.1.5 from FTP
+ *
+/* Revision 1.2  2000/01/05 08:20:39  markster
+/* Some OSS fixes and a few lpc changes to make it actually work
+/*
+ * Revision 1.2  1996/08/20  20:42:59  jaf
+ * Removed all static local variables that were SAVE'd in the Fortran
+ * code, and put them in struct lpc10_decoder_state that is passed as an
+ * argument.
+ *
+ * Removed init function, since all initialization is now done in
+ * init_lpc10_decoder_state().
+ *
+ * Revision 1.1  1996/08/19  22:30:33  jaf
+ * Initial revision
+ * */
+/* Revision 1.3  1996/03/29  22:03:47  jaf */
+/* Removed definitions for any constants that were no longer used. */
+
+/* Revision 1.2  1996/03/26  19:34:33  jaf */
+/* Added comments indicating which constants are not needed in an */
+/* application that uses the LPC-10 coder. */
+
+/* Revision 1.1  1996/02/07  14:43:51  jaf */
+/* Initial revision */
+
+/*   LPC Configuration parameters: */
+/* Frame size, Prediction order, Pitch period */
+/*       Arguments */
+/* $Log$
+ * Revision 1.4  2000/01/05 08:20:39  markster
+ * Version 0.1.5 from FTP
+ *
+/* Revision 1.2  2000/01/05 08:20:39  markster
+/* Some OSS fixes and a few lpc changes to make it actually work
+/*
+ * Revision 1.2  1996/08/20  20:42:59  jaf
+ * Removed all static local variables that were SAVE'd in the Fortran
+ * code, and put them in struct lpc10_decoder_state that is passed as an
+ * argument.
+ *
+ * Removed init function, since all initialization is now done in
+ * init_lpc10_decoder_state().
+ *
+ * Revision 1.1  1996/08/19  22:30:33  jaf
+ * Initial revision
+ * */
+/* Revision 1.3  1996/03/29  22:05:55  jaf */
+/* Commented out the common block variables that are not needed by the */
+/* embedded version. */
+
+/* Revision 1.2  1996/03/26  19:34:50  jaf */
+/* Added comments indicating which constants are not needed in an */
+/* application that uses the LPC-10 coder. */
+
+/* Revision 1.1  1996/02/07  14:44:09  jaf */
+/* Initial revision */
+
+/*   LPC Processing control variables: */
+
+/* *** Read-only: initialized in setup */
+
+/*  Files for Speech, Parameter, and Bitstream Input & Output, */
+/*    and message and debug outputs. */
+
+/* Here are the only files which use these variables: */
+
+/* lpcsim.f setup.f trans.f error.f vqsetup.f */
+
+/* Many files which use fdebug are not listed, since it is only used in */
+/* those other files conditionally, to print trace statements. */
+/* 	integer fsi, fso, fpi, fpo, fbi, fbo, pbin, fmsg, fdebug */
+/*  LPC order, Frame size, Quantization rate, Bits per frame, */
+/*    Error correction */
+/* Subroutine SETUP is the only place where order is assigned a value, */
+/* and that value is 10.  It could increase efficiency 1% or so to */
+/* declare order as a constant (i.e., a Fortran PARAMETER) instead of as 
+*/
+/* a variable in a COMMON block, since it is used in many places in the */
+/* core of the coding and decoding routines.  Actually, I take that back. 
+*/
+/* At least when compiling with f2c, the upper bound of DO loops is */
+/* stored in a local variable before the DO loop begins, and then that is 
+*/
+/* compared against on each iteration. */
+/* Similarly for lframe, which is given a value of MAXFRM in SETUP. */
+/* Similarly for quant, which is given a value of 2400 in SETUP.  quant */
+/* is used in only a few places, and never in the core coding and */
+/* decoding routines, so it could be eliminated entirely. */
+/* nbits is similar to quant, and is given a value of 54 in SETUP. */
+/* corrp is given a value of .TRUE. in SETUP, and is only used in the */
+/* subroutines ENCODE and DECODE.  It doesn't affect the speed of the */
+/* coder significantly whether it is .TRUE. or .FALSE., or whether it is 
+*/
+/* a constant or a variable, since it is only examined once per frame. */
+/* Leaving it as a variable that is set to .TRUE.  seems like a good */
+/* idea, since it does enable some error-correction capability for */
+/* unvoiced frames, with no change in the coding rate, and no noticeable 
+*/
+/* quality difference in the decoded speech. */
+/* 	integer quant, nbits */
+/* *** Read/write: variables for debugging, not needed for LPC algorithm 
+*/
+
+/*  Current frame, Unstable frames, Output clip count, Max onset buffer, 
+*/
+/*    Debug listing detail level, Line count on listing page */
+
+/* nframe is not needed for an embedded LPC10 at all. */
+/* nunsfm is initialized to 0 in SETUP, and incremented in subroutine */
+/* ERROR, which is only called from RCCHK.  When LPC10 is embedded into */
+/* an application, I would recommend removing the call to ERROR in RCCHK, 
+*/
+/* and remove ERROR and nunsfm completely. */
+/* iclip is initialized to 0 in SETUP, and incremented in entry SWRITE in 
+*/
+/* sread.f.  When LPC10 is embedded into an application, one might want */
+/* to cause it to be incremented in a routine that takes the output of */
+/* SYNTHS and sends it to an audio device.  It could be optionally */
+/* displayed, for those that might want to know what it is. */
+/* maxosp is never initialized to 0 in SETUP, although it probably should 
+*/
+/* be, and it is updated in subroutine ANALYS.  I doubt that its value */
+/* would be of much interest to an application in which LPC10 is */
+/* embedded. */
+/* listl and lincnt are not needed for an embedded LPC10 at all. */
+/* 	integer nframe, nunsfm, iclip, maxosp, listl, lincnt */
+/* 	common /contrl/ fsi, fso, fpi, fpo, fbi, fbo, pbin, fmsg, fdebug */
+/* 	common /contrl/ quant, nbits */
+/* 	common /contrl/ nframe, nunsfm, iclip, maxosp, listl, lincnt */
+/*       Parameters/constants */
+/*       Local variables that need not be saved */
+/*       Local state */
+/*       BUF is a buffer of speech samples that would have been returned 
+*/
+/*       by the older version of SYNTHS, but the newer version doesn't, */
+/*       so that the newer version can always return MAXFRM samples on */
+/*       every call.  This has the effect of delaying the return of */
+/*       samples for one additional frame time. */
+
+/*       Indices 1 through BUFLEN contain samples that are left over from 
+*/
+/*       the last call to SYNTHS.  Given the way that PITSYN works, */
+/*       BUFLEN should always be in the range MAXFRM-MAXPIT+1 through */
+/*       MAXFRM, inclusive, after a call to SYNTHS is complete. */
+
+/*       On the first call to SYNTHS (or the first call after */
+/*       reinitializing with the entry INITSYNTHS), BUFLEN is MAXFRM, and 
+*/
+/*       a frame of silence is always returned. */
+    /* Parameter adjustments */
+    if (voice) {
+	--voice;
+	}
+    if (rc) {
+	--rc;
+	}
+    if (speech) {
+	--speech;
+	}
+
+    /* Function Body */
+    buf = &(st->buf[0]);
+    buflen = &(st->buflen);
+
+/* Computing MAX */
+    i__1 = min(*pitch,156);
+    *pitch = max(i__1,20);
+    i__1 = contrl_1.order;
+    for (i__ = 1; i__ <= i__1; ++i__) {
+/* Computing MAX */
+/* Computing MIN */
+	r__2 = rc[i__];
+	r__1 = min(r__2,.99f);
+	rc[i__] = max(r__1,-.99f);
+    }
+    pitsyn_(&contrl_1.order, &voice[1], pitch, rms, &rc[1], &contrl_1.lframe, 
+	    ivuv, ipiti, rmsi, rci, &nout, &ratio, st);
+    if (nout > 0) {
+	i__1 = nout;
+	for (j = 1; j <= i__1; ++j) {
+
+/*             Add synthesized speech for pitch period J to the en
+d of */
+/*             BUF. */
+
+	    irc2pc_(&rci[j * 10 - 10], pc, &contrl_1.order, &c_b2, &g2pass);
+	    bsynz_(pc, &ipiti[j - 1], &ivuv[j - 1], &buf[*buflen], &rmsi[j - 1]
+		    , &ratio, &g2pass, st);
+	    deemp_(&buf[*buflen], &ipiti[j - 1], st);
+	    *buflen += ipiti[j - 1];
+	}
+
+/*          Copy first MAXFRM samples from BUF to output array SPEECH 
+*/
+/*          (scaling them), and then remove them from the beginning of
+ */
+/*          BUF. */
+
+	for (i__ = 1; i__ <= 180; ++i__) {
+	    speech[i__] = buf[i__ - 1] / 4096.f;
+	}
+	*k = 180;
+	*buflen += -180;
+	i__1 = *buflen;
+	for (i__ = 1; i__ <= i__1; ++i__) {
+	    buf[i__ - 1] = buf[i__ + 179];
+	}
+    }
+    return 0;
+} /* synths_ */

codecs/lpc10/tbdm.c

@@ -0,0 +1,176 @@
+/*
+
+$Log$
+Revision 1.4  2000/01/05 08:20:40  markster
+Version 0.1.5 from FTP
+
+Revision 1.2  2000/01/05 08:20:40  markster
+Some OSS fixes and a few lpc changes to make it actually work
+
+ * Revision 1.1  1996/08/19  22:30:26  jaf
+ * Initial revision
+ *
+
+*/
+
+#ifdef P_R_O_T_O_T_Y_P_E_S
+extern int tbdm_(real *speech, integer *lpita, integer *tau, integer *ltau, real *amdf, integer *minptr, integer *maxptr, integer *mintau);
+/*:ref: difmag_ 14 8 6 4 4 4 4 6 4 4 */
+#endif
+
+/*  -- translated by f2c (version 19951025).
+   You must link the resulting object file with the libraries:
+	-lf2c -lm   (in that order)
+*/
+
+#include "f2c.h"
+
+/* ********************************************************************** */
+
+/* 	TBDM Version 49 */
+
+/* $Log$
+ * Revision 1.4  2000/01/05 08:20:40  markster
+ * Version 0.1.5 from FTP
+ *
+/* Revision 1.2  2000/01/05 08:20:40  markster
+/* Some OSS fixes and a few lpc changes to make it actually work
+/*
+ * Revision 1.1  1996/08/19  22:30:26  jaf
+ * Initial revision
+ * */
+/* Revision 1.3  1996/03/18  22:14:00  jaf */
+/* Just added a few comments about which array indices of the arguments */
+/* are used, and mentioning that this subroutine has no local state. */
+
+/* Revision 1.2  1996/03/13  14:48:37  jaf */
+/* Comments added explaining that none of the local variables of this */
+/* subroutine need to be saved from one invocation to the next. */
+
+/* Revision 1.1  1996/02/07 14:49:54  jaf */
+/* Initial revision */
+
+
+/* ********************************************************************* */
+
+/*TURBO DIFMAG: Compute High Resolution Average Magnitude Difference Function
+*/
+
+/* Note: There are several constants in here that appear to depend on a */
+/* particular TAU table.  That's not a problem for the LPC10 coder, but */
+/* watch out if you change the contents of TAU in the subroutine ANALYS. */
+
+/* Input: */
+/*  SPEECH - Low pass filtered speech */
+/*           Indices 1 through MAX+LPITA-1 are read, where: */
+/*           MAX = (TAU(LTAU)-TAU(1))/2+1 */
+/*           (If TAU(1) .LT. 39, then larger indices could be read */
+/*           by the last call to DIFMAG below.) */
+/*  LPITA  - Length of speech buffer */
+/*  TAU    - Table of lags, sorted in increasing order. */
+/*           Indices 1 through LTAU read. */
+/*  LTAU   - Number of lag values to compute */
+/* Output: */
+/*  AMDF   - Average Magnitude Difference for each lag in TAU */
+/*          Indices 1 through LTAU written, and several might then be read.*/
+/*  MINPTR - Index of minimum AMDF value */
+/*  MAXPTR - Index of maximum AMDF value within +/- 1/2 octave of min */
+/*  MINTAU - Lag corresponding to minimum AMDF value */
+
+/* This subroutine has no local state. */
+
+/* Subroutine */ int tbdm_(real *speech, integer *lpita, integer *tau, 
+	integer *ltau, real *amdf, integer *minptr, integer *maxptr, integer *
+	mintau)
+{
+    /* System generated locals */
+    integer i__1, i__2, i__3, i__4;
+
+    /* Local variables */
+    real amdf2[6];
+    integer minp2, ltau2, maxp2, i__;
+    extern /* Subroutine */ int difmag_(real *, integer *, integer *, integer 
+	    *, integer *, real *, integer *, integer *);
+    integer minamd, ptr, tau2[6];
+
+/* 	Arguments */
+/* 	REAL SPEECH(LPITA+TAU(LTAU)), AMDF(LTAU) */
+/*   Stupid TOAST doesn't understand expressions */
+/*       Local variables that need not be saved */
+/*       Local state */
+/*       None */
+/*   Compute full AMDF using log spaced lags, find coarse minimum */
+    /* Parameter adjustments */
+    --speech;
+    --amdf;
+    --tau;
+
+    /* Function Body */
+    difmag_(&speech[1], lpita, &tau[1], ltau, &tau[*ltau], &amdf[1], minptr, 
+	    maxptr);
+    *mintau = tau[*minptr];
+    minamd = amdf[*minptr];
+/*   Build table containing all lags within +/- 3 of the AMDF minimum */
+/*    excluding all that have already been computed */
+    ltau2 = 0;
+    ptr = *minptr - 2;
+/* Computing MAX */
+    i__1 = *mintau - 3;
+/* Computing MIN */
+    i__3 = *mintau + 3, i__4 = tau[*ltau] - 1;
+    i__2 = min(i__3,i__4);
+    for (i__ = max(i__1,41); i__ <= i__2; ++i__) {
+	while(tau[ptr] < i__) {
+	    ++ptr;
+	}
+	if (tau[ptr] != i__) {
+	    ++ltau2;
+	    tau2[ltau2 - 1] = i__;
+	}
+    }
+/*   Compute AMDF of the new lags, if there are any, and choose one */
+/*    if it is better than the coarse minimum */
+    if (ltau2 > 0) {
+	difmag_(&speech[1], lpita, tau2, &ltau2, &tau[*ltau], amdf2, &minp2, &
+		maxp2);
+	if (amdf2[minp2 - 1] < (real) minamd) {
+	    *mintau = tau2[minp2 - 1];
+	    minamd = amdf2[minp2 - 1];
+	}
+    }
+/*   Check one octave up, if there are any lags not yet computed */
+    if (*mintau >= 80) {
+	i__ = *mintau / 2;
+	if ((i__ & 1) == 0) {
+	    ltau2 = 2;
+	    tau2[0] = i__ - 1;
+	    tau2[1] = i__ + 1;
+	} else {
+	    ltau2 = 1;
+	    tau2[0] = i__;
+	}
+	difmag_(&speech[1], lpita, tau2, &ltau2, &tau[*ltau], amdf2, &minp2, &
+		maxp2);
+	if (amdf2[minp2 - 1] < (real) minamd) {
+	    *mintau = tau2[minp2 - 1];
+	    minamd = amdf2[minp2 - 1];
+	    *minptr += -20;
+	}
+    }
+/*   Force minimum of the AMDF array to the high resolution minimum */
+    amdf[*minptr] = (real) minamd;
+/*   Find maximum of AMDF within 1/2 octave of minimum */
+/* Computing MAX */
+    i__2 = *minptr - 5;
+    *maxptr = max(i__2,1);
+/* Computing MIN */
+    i__1 = *minptr + 5;
+    i__2 = min(i__1,*ltau);
+    for (i__ = *maxptr + 1; i__ <= i__2; ++i__) {
+	if (amdf[i__] > amdf[*maxptr]) {
+	    *maxptr = i__;
+	}
+    }
+    return 0;
+} /* tbdm_ */
+

codecs/lpc10/voicin.c

@@ -0,0 +1,759 @@
+/*
+
+$Log$
+Revision 1.4  2000/01/05 08:20:40  markster
+Version 0.1.5 from FTP
+
+Revision 1.2  2000/01/05 08:20:40  markster
+Some OSS fixes and a few lpc changes to make it actually work
+
+ * Revision 1.2  1996/08/20  20:45:00  jaf
+ * Removed all static local variables that were SAVE'd in the Fortran
+ * code, and put them in struct lpc10_encoder_state that is passed as an
+ * argument.
+ *
+ * Removed init function, since all initialization is now done in
+ * init_lpc10_encoder_state().
+ *
+ * Revision 1.1  1996/08/19  22:30:14  jaf
+ * Initial revision
+ *
+
+*/
+
+#ifdef P_R_O_T_O_T_Y_P_E_S
+extern int voicin_(integer *vwin, real *inbuf, real *lpbuf, integer *buflim, integer *half, real *minamd, real *maxamd, integer *mintau, real *ivrc, integer *obound, integer *voibuf, integer *af, struct lpc10_encoder_state *st);
+/* comlen contrl_ 12 */
+/*:ref: vparms_ 14 14 4 6 6 4 4 6 4 4 4 4 6 6 6 6 */
+#endif
+
+/*  -- translated by f2c (version 19951025).
+   You must link the resulting object file with the libraries:
+	-lf2c -lm   (in that order)
+*/
+
+#include "f2c.h"
+
+/* Common Block Declarations */
+
+extern struct {
+    integer order, lframe;
+    logical corrp;
+} contrl_;
+
+#define contrl_1 contrl_
+
+/****************************************************************************/
+
+/* 	VOICIN Version 52 */
+
+/* $Log$
+ * Revision 1.4  2000/01/05 08:20:40  markster
+ * Version 0.1.5 from FTP
+ *
+/* Revision 1.2  2000/01/05 08:20:40  markster
+/* Some OSS fixes and a few lpc changes to make it actually work
+/*
+ * Revision 1.2  1996/08/20  20:45:00  jaf
+ * Removed all static local variables that were SAVE'd in the Fortran
+ * code, and put them in struct lpc10_encoder_state that is passed as an
+ * argument.
+ *
+ * Removed init function, since all initialization is now done in
+ * init_lpc10_encoder_state().
+ *
+ * Revision 1.1  1996/08/19  22:30:14  jaf
+ * Initial revision
+ * */
+/* Revision 1.10  1996/03/29  17:59:14  jaf */
+/* Avoided using VALUE(9), although it shouldn't affect the function of */
+/* the code at all, because it was always multiplied by VDC(9,SNRL), */
+/* which is 0 for all values of SNRL.  Still, if VALUE(9) had an initial */
+/* value of IEEE NaN, it might cause trouble (I don't know how IEEE */
+/* defines Nan * 0.  It should either be NaN or 0.) */
+
+/* Revision 1.9  1996/03/29  17:54:46  jaf */
+/* Added a few comments about the accesses made to argument array VOIBUF */
+/* and the local saved array VOICE. */
+
+/* Revision 1.8  1996/03/27  18:19:54  jaf */
+/* Added an assignment to VSTATE that does not affect the function of the */
+/* program at all.  The only reason I put it in was so that the tracing */
+/* statements at the end, when enabled, will print a consistent value for */
+/* VSTATE when HALF .EQ. 1, rather than a garbage value that could change */
+/* from one call to the next. */
+
+/* Revision 1.7  1996/03/26  20:00:06  jaf */
+/* Removed the inclusion of the file "vcomm.fh", and put its contents */
+/* into this file.  It was included nowhere else but here. */
+
+/* Revision 1.6  1996/03/26  19:38:09  jaf */
+/* Commented out trace statements. */
+
+/* Revision 1.5  1996/03/19  20:43:45  jaf */
+/* Added comments about which indices of OBOUND and VOIBUF can be */
+/* accessed, and whether they are read or written.  VOIBUF is fairly */
+/* messy. */
+
+/* Revision 1.4  1996/03/19  15:00:58  jaf */
+/* Moved the DATA statements for the *VDC* variables later, as it is */
+/* apparently illegal to have DATA statements before local variable */
+/* declarations. */
+
+/* Revision 1.3  1996/03/19  00:10:49  jaf */
+/* Heavily commented the local variables that are saved from one */
+/* invocation to the next, and how the local variable FIRST is used to */
+/* avoid the need to assign most of them initial values with DATA */
+/* statements. */
+
+/* A few should be initialized, but aren't.  I've guessed initial values */
+/* for two of these, SFBUE and SLBUE, and I've convinced myself that for */
+/* VOICE, the effects of uninitialized values will die out after 2 or 3 */
+/* frame times.  It would still be good to choose initial values for */
+/* these, but I don't know what reasonable values would be (0 comes to */
+/* mind). */
+
+/* Revision 1.2  1996/03/13  16:09:28  jaf */
+/* Comments added explaining which of the local variables of this */
+/* subroutine need to be saved from one invocation to the next, and which */
+/* do not. */
+
+/* WARNING!  Some of them that should are never given initial values in */
+/* this code.  Hopefully, Fortran 77 defines initial values for them, but */
+/* even so, giving them explicit initial values is preferable. */
+
+/* WARNING!  VALUE(9) is used, but never assigned a value.  It should */
+/* probably be eliminated from the code. */
+
+/* Revision 1.1  1996/02/07 14:50:28  jaf */
+/* Initial revision */
+
+
+/****************************************************************************/
+
+/*        Voicing Detection (VOICIN) makes voicing decisions for each half */
+/*  frame of input speech.  Tentative voicing decisions are made two frames*/
+/*   in the future (2F) for each half frame.  These decisions are carried */
+/*   through one frame in the future (1F) to the present (P) frame where */
+/*   they are examined and smoothed, resulting in the final voicing */
+/*   decisions for each half frame. */
+/*        The voicing parameter (signal measurement) column vector (VALUE) */
+/*   is based on a rectangular window of speech samples determined by the */
+/*  window placement algorithm.  The voicing parameter vector contains the*/
+/*  AMDF windowed maximum-to-minimum ratio, the zero crossing rate, energy*/
+/*   measures, reflection coefficients, and prediction gains.  The voicing */
+/*  window is placed to avoid contamination of the voicing parameter vector*/
+/*   with speech onsets. */
+/*        The input signal is then classified as unvoiced (including */
+/*   silence) or voiced.  This decision is made by a linear discriminant */
+/*   function consisting of a dot product of the voicing decision */
+/*   coefficient (VDC) row vector with the measurement column vector */
+/*  (VALUE).  The VDC vector is 2-dimensional, each row vector is optimized*/
+/*   for a particular signal-to-noise ratio (SNR).  So, before the dot */
+/*   product is performed, the SNR is estimated to select the appropriate */
+/*   VDC vector. */
+/*        The smoothing algorithm is a modified median smoother.  The */
+/*  voicing discriminant function is used by the smoother to determine how*/
+/*   strongly voiced or unvoiced a signal is.  The smoothing is further */
+/*   modified if a speech onset and a voicing decision transition occur */
+/*   within one half frame.  In this case, the voicing decision transition */
+/*  is extended to the speech onset.  For transmission purposes, there are*/
+/*   constraints on the duration and transition of voicing decisions.  The */
+/*   smoother takes these constraints into account. */
+/*        Finally, the energy estimates are updated along with the dither */
+/*   threshold used to calculate the zero crossing rate (ZC). */
+
+/* Inputs: */
+/*  VWIN      - Voicing window limits */
+/*              The indices read of arrays VWIN, INBUF, LPBUF, and BUFLIM */
+/*              are the same as those read by subroutine VPARMS. */
+/*  INBUF     - Input speech buffer */
+/*  LPBUF     - Low-pass filtered speech buffer */
+/*  BUFLIM    - INBUF and LPBUF limits */
+/*  HALF      - Present analysis half frame number */
+/*  MINAMD    - Minimum value of the AMDF */
+/*  MAXAMD    - Maximum value of the AMDF */
+/*  MINTAU    - Pointer to the lag of the minimum AMDF value */
+/*  IVRC(2)   - Inverse filter's RC's */
+/*              Only index 2 of array IVRC read under normal operation. */
+/*              (Index 1 is also read when debugging is turned on.) */
+/*  OBOUND    - Onset boundary descriptions */
+/*             Indices 1 through 3 read if (HALF .NE. 1), otherwise untouched.
+*/
+/*  AF        - The analysis frame number */
+/* Output: */
+/*  VOIBUF(2,0:AF) - Buffer of voicing decisions */
+/*              Index (HALF,3) written. */
+/*              If (HALF .EQ. 1), skip down to "Read (HALF,3)" below. */
+/*              Indices (1,2), (2,1), (1,2), and (2,2) read. */
+/*              One of the following is then done: */
+/*                 read (1,3) and possibly write (1,2) */
+/*                 read (1,3) and write (1,2) or (2,2) */
+/*                 write (2,1) */
+/*                 write (2,1) or (1,2) */
+/*                 read (1,0) and (1,3) and then write (2,2) or (1,1) */
+/*                 no reads or writes on VOIBUF */
+/*              Finally, read (HALF,3) */
+/* Internal: */
+/*  QS        - Ratio of preemphasized to full-band energies */
+/*  RC1       - First reflection coefficient */
+/* AR_B      - Product of the causal forward and reverse pitch prediction gain
+s*/
+/* AR_F      - Product of the noncausal forward and rev. pitch prediction gain
+s*/
+/*  ZC        - Zero crossing rate */
+/*  DITHER    - Zero crossing threshold level */
+/*  MAXMIN    - AMDF's 1 octave windowed maximum-to-minimum ratio */
+/*  MINPTR    - Location  of minimum AMDF value */
+/*  NVDC      - Number of elements in each VDC vector */
+/*  NVDCL     - Number of VDC vectors */
+/*  VDCL      - SNR values corresponding to the set of VDC's */
+/*  VDC       - 2-D voicing decision coefficient vector */
+/*  VALUE(9)  - Voicing Parameters */
+/*  VOICE(2,3)- History of LDA results */
+/*              On every call when (HALF .EQ. 1), VOICE(*,I+1) is */
+/*              shifted back to VOICE(*,I), for I=1,2. */
+/*              VOICE(HALF,3) is written on every call. */
+/*              Depending on several conditions, one or more of */
+/*              (1,1), (1,2), (2,1), and (2,2) might then be read. */
+/*  LBE       - Ratio of low-band instantaneous to average energies */
+/*  FBE       - Ratio of full-band instantaneous to average energies */
+/*  LBVE      - Low band voiced energy */
+/*  LBUE      - Low band unvoiced energy */
+/*  FBVE      - Full band voiced energy */
+/*  FBUE      - Full band unvoiced energy */
+/*  OFBUE     - Previous full-band unvoiced energy */
+/*  OLBUE     - Previous low-band unvoiced energy */
+/*  REF       - Reference energy for initialization and DITHER threshold */
+/*  SNR       - Estimate of signal-to-noise ratio */
+/*  SNR2      - Estimate of low-band signal-to-noise ratio */
+/*  SNRL      - SNR level number */
+/*  OT        - Onset transition present */
+/*  VSTATE    - Decimal interpretation of binary voicing classifications */
+/*  FIRST     - First call flag */
+
+/* This subroutine maintains local state from one call to the next.  If */
+/* you want to switch to using a new audio stream for this filter, or */
+/* reinitialize its state for any other reason, call the ENTRY */
+/* INITVOICIN. */
+
+/* Subroutine */ int voicin_(integer *vwin, real *inbuf, real *
+	lpbuf, integer *buflim, integer *half, real *minamd, real *maxamd, 
+	integer *mintau, real *ivrc, integer *obound, integer *voibuf, 
+	integer *af, struct lpc10_encoder_state *st)
+{
+    /* Initialized data */
+
+    real *dither;
+    static real vdc[100]	/* was [10][10] */ = { 0.f,1714.f,-110.f,
+	    334.f,-4096.f,-654.f,3752.f,3769.f,0.f,1181.f,0.f,874.f,-97.f,
+	    300.f,-4096.f,-1021.f,2451.f,2527.f,0.f,-500.f,0.f,510.f,-70.f,
+	    250.f,-4096.f,-1270.f,2194.f,2491.f,0.f,-1500.f,0.f,500.f,-10.f,
+	    200.f,-4096.f,-1300.f,2e3f,2e3f,0.f,-2e3f,0.f,500.f,0.f,0.f,
+	    -4096.f,-1300.f,2e3f,2e3f,0.f,-2500.f,0.f,0.f,0.f,0.f,0.f,0.f,0.f,
+	    0.f,0.f,0.f,0.f,0.f,0.f,0.f,0.f,0.f,0.f,0.f,0.f,0.f,0.f,0.f,0.f,
+	    0.f,0.f,0.f,0.f,0.f,0.f,0.f,0.f,0.f,0.f,0.f,0.f,0.f,0.f,0.f,0.f,
+	    0.f,0.f,0.f,0.f,0.f,0.f,0.f,0.f,0.f,0.f,0.f };
+    static integer nvdcl = 5;
+    static real vdcl[10] = { 600.f,450.f,300.f,200.f,0.f,0.f,0.f,0.f,0.f,0.f }
+	    ;
+
+    /* System generated locals */
+    integer inbuf_offset, lpbuf_offset, i__1, i__2;
+    real r__1, r__2;
+
+    /* Builtin functions */
+    integer i_nint(real *);
+    double sqrt(doublereal);
+
+    /* Local variables */
+    real ar_b__, ar_f__;
+    integer *lbve, *lbue, *fbve, *fbue;
+    integer snrl, i__;
+    integer *ofbue, *sfbue;
+    real *voice;
+    integer *olbue, *slbue;
+    real value[9];
+    integer zc;
+    logical ot;
+    real qs;
+    real *maxmin;
+    integer vstate;
+    real rc1;
+    extern /* Subroutine */ int vparms_(integer *, real *, real *, integer *, 
+	    integer *, real *, integer *, integer *, integer *, integer *, 
+	    real *, real *, real *, real *);
+    integer fbe, lbe;
+    real *snr;
+    real snr2;
+
+/* 	Global Variables: */
+/*       Arguments */
+/* $Log$
+ * Revision 1.4  2000/01/05 08:20:40  markster
+ * Version 0.1.5 from FTP
+ *
+/* Revision 1.2  2000/01/05 08:20:40  markster
+/* Some OSS fixes and a few lpc changes to make it actually work
+/*
+ * Revision 1.2  1996/08/20  20:45:00  jaf
+ * Removed all static local variables that were SAVE'd in the Fortran
+ * code, and put them in struct lpc10_encoder_state that is passed as an
+ * argument.
+ *
+ * Removed init function, since all initialization is now done in
+ * init_lpc10_encoder_state().
+ *
+ * Revision 1.1  1996/08/19  22:30:14  jaf
+ * Initial revision
+ * */
+/* Revision 1.3  1996/03/29  22:05:55  jaf */
+/* Commented out the common block variables that are not needed by the */
+/* embedded version. */
+
+/* Revision 1.2  1996/03/26  19:34:50  jaf */
+/* Added comments indicating which constants are not needed in an */
+/* application that uses the LPC-10 coder. */
+
+/* Revision 1.1  1996/02/07  14:44:09  jaf */
+/* Initial revision */
+
+/*   LPC Processing control variables: */
+
+/* *** Read-only: initialized in setup */
+
+/*  Files for Speech, Parameter, and Bitstream Input & Output, */
+/*    and message and debug outputs. */
+
+/* Here are the only files which use these variables: */
+
+/* lpcsim.f setup.f trans.f error.f vqsetup.f */
+
+/* Many files which use fdebug are not listed, since it is only used in */
+/* those other files conditionally, to print trace statements. */
+/* 	integer fsi, fso, fpi, fpo, fbi, fbo, pbin, fmsg, fdebug */
+/*  LPC order, Frame size, Quantization rate, Bits per frame, */
+/*    Error correction */
+/* Subroutine SETUP is the only place where order is assigned a value, */
+/* and that value is 10.  It could increase efficiency 1% or so to */
+/* declare order as a constant (i.e., a Fortran PARAMETER) instead of as 
+*/
+/* a variable in a COMMON block, since it is used in many places in the */
+/* core of the coding and decoding routines.  Actually, I take that back. 
+*/
+/* At least when compiling with f2c, the upper bound of DO loops is */
+/* stored in a local variable before the DO loop begins, and then that is 
+*/
+/* compared against on each iteration. */
+/* Similarly for lframe, which is given a value of MAXFRM in SETUP. */
+/* Similarly for quant, which is given a value of 2400 in SETUP.  quant */
+/* is used in only a few places, and never in the core coding and */
+/* decoding routines, so it could be eliminated entirely. */
+/* nbits is similar to quant, and is given a value of 54 in SETUP. */
+/* corrp is given a value of .TRUE. in SETUP, and is only used in the */
+/* subroutines ENCODE and DECODE.  It doesn't affect the speed of the */
+/* coder significantly whether it is .TRUE. or .FALSE., or whether it is 
+*/
+/* a constant or a variable, since it is only examined once per frame. */
+/* Leaving it as a variable that is set to .TRUE.  seems like a good */
+/* idea, since it does enable some error-correction capability for */
+/* unvoiced frames, with no change in the coding rate, and no noticeable 
+*/
+/* quality difference in the decoded speech. */
+/* 	integer quant, nbits */
+/* *** Read/write: variables for debugging, not needed for LPC algorithm 
+*/
+
+/*  Current frame, Unstable frames, Output clip count, Max onset buffer, 
+*/
+/*    Debug listing detail level, Line count on listing page */
+
+/* nframe is not needed for an embedded LPC10 at all. */
+/* nunsfm is initialized to 0 in SETUP, and incremented in subroutine */
+/* ERROR, which is only called from RCCHK.  When LPC10 is embedded into */
+/* an application, I would recommend removing the call to ERROR in RCCHK, 
+*/
+/* and remove ERROR and nunsfm completely. */
+/* iclip is initialized to 0 in SETUP, and incremented in entry SWRITE in 
+*/
+/* sread.f.  When LPC10 is embedded into an application, one might want */
+/* to cause it to be incremented in a routine that takes the output of */
+/* SYNTHS and sends it to an audio device.  It could be optionally */
+/* displayed, for those that might want to know what it is. */
+/* maxosp is never initialized to 0 in SETUP, although it probably should 
+*/
+/* be, and it is updated in subroutine ANALYS.  I doubt that its value */
+/* would be of much interest to an application in which LPC10 is */
+/* embedded. */
+/* listl and lincnt are not needed for an embedded LPC10 at all. */
+/* 	integer nframe, nunsfm, iclip, maxosp, listl, lincnt */
+/* 	common /contrl/ fsi, fso, fpi, fpo, fbi, fbo, pbin, fmsg, fdebug */
+/* 	common /contrl/ quant, nbits */
+/* 	common /contrl/ nframe, nunsfm, iclip, maxosp, listl, lincnt */
+/* 	Parameters/constants */
+/*       Voicing coefficient and Linear Discriminant Analysis variables: 
+*/
+/*       Max number of VDC's and VDC levels */
+/*       The following are not Fortran PARAMETER's, but they are */
+/*       initialized with DATA statements, and never modified. */
+/*       Actual number of VDC's and levels */
+/*       Local variables that need not be saved */
+/*       Note: */
+
+/*       VALUE(1) through VALUE(8) are assigned values, but VALUE(9) */
+/*       never is.  Yet VALUE(9) is read in the loop that begins "DO I = 
+*/
+/*       1, 9" below.  I believe that this doesn't cause any problems in 
+*/
+/*       this subroutine, because all VDC(9,*) array elements are 0, and 
+*/
+/*       this is what is multiplied by VALUE(9) in all cases.  Still, it 
+*/
+/*       would save a multiplication to change the loop to "DO I = 1, 8". 
+*/
+/*       Local state */
+/*       WARNING! */
+
+/*       VOICE, SFBUE, and SLBUE should be saved from one invocation to */
+/*       the next, but they are never given an initial value. */
+
+/*       Does Fortran 77 specify some default initial value, like 0, or */
+/*       is it undefined?  If it is undefined, then this code should be */
+/*       corrected to specify an initial value. */
+
+/*       For VOICE, note that it is "shifted" in the statement that */
+/*       begins "IF (HALF .EQ. 1) THEN" below.  Also, uninitialized */
+/*       values in the VOICE array can only affect entries in the VOIBUF 
+*/
+/*       array that are for the same frame, or for an older frame.  Thus 
+*/
+/*       the effects of uninitialized values in VOICE cannot linger on */
+/*       for more than 2 or 3 frame times. */
+
+/*       For SFBUE and SLBUE, the effects of uninitialized values can */
+/*       linger on for many frame times, because their previous values */
+/*       are exponentially decayed.  Thus it is more important to choose 
+*/
+/*       initial values for these variables.  I would guess that a */
+/*       reasonable initial value for SFBUE is REF/16, the same as used */
+/*       for FBUE and OFBUE.  Similarly, SLBUE can be initialized to */
+/*       REF/32, the same as for LBUE and OLBUE. */
+
+/*       These guessed initial values should be validated by re-running */
+/*       the modified program on some audio samples. */
+
+/*   Declare and initialize filters: */
+
+    dither = (&st->dither);
+    snr = (&st->snr);
+    maxmin = (&st->maxmin);
+    voice = (&st->voice[0]);
+    lbve = (&st->lbve);
+    lbue = (&st->lbue);
+    fbve = (&st->fbve);
+    fbue = (&st->fbue);
+    ofbue = (&st->ofbue);
+    olbue = (&st->olbue);
+    sfbue = (&st->sfbue);
+    slbue = (&st->slbue);
+
+    /* Parameter adjustments */
+    if (vwin) {
+	--vwin;
+	}
+    if (buflim) {
+	--buflim;
+	}
+    if (inbuf) {
+	inbuf_offset = buflim[1];
+	inbuf -= inbuf_offset;
+	}
+    if (lpbuf) {
+	lpbuf_offset = buflim[3];
+	lpbuf -= lpbuf_offset;
+	}
+    if (ivrc) {
+	--ivrc;
+	}
+    if (obound) {
+	--obound;
+	}
+    if (voibuf) {
+	--voibuf;
+	}
+
+    /* Function Body */
+
+/*       The following variables are saved from one invocation to the */
+/*       next, but are not initialized with DATA statements.  This is */
+/*       acceptable, because FIRST is initialized ot .TRUE., and the */
+/*       first time that this subroutine is then called, they are all */
+/*       given initial values. */
+
+/*       SNR */
+/*       LBVE, LBUE, FBVE, FBUE, OFBUE, OLBUE */
+
+/*       MAXMIN is initialized on the first call, assuming that HALF */
+/*       .EQ. 1 on first call.  This is how ANALYS calls this subroutine. 
+*/
+
+/*   Voicing Decision Parameter vector (* denotes zero coefficient): */
+
+/* 	* MAXMIN */
+/* 	  LBE/LBVE */
+/* 	  ZC */
+/* 	  RC1 */
+/* 	  QS */
+/* 	  IVRC2 */
+/* 	  aR_B */
+/* 	  aR_F */
+/* 	* LOG(LBE/LBVE) */
+/*  Define 2-D voicing decision coefficient vector according to the voicin
+g*/
+/*  parameter order above.  Each row (VDC vector) is optimized for a speci
+fic*/
+/*   SNR.  The last element of the vector is the constant. */
+/* 	         E    ZC    RC1    Qs   IVRC2  aRb   aRf        c */
+
+/*  The VOICE array contains the result of the linear discriminant functio
+n*/
+/*   (analog values).  The VOIBUF array contains the hard-limited binary 
+*/
+/*   voicing decisions.  The VOICE and VOIBUF arrays, according to FORTRAN
+ */
+/*   memory allocation, are addressed as: */
+
+/* 	   (half-frame number, future-frame number) */
+
+/* 	   |   Past    |  Present  |  Future1  |  Future2  | */
+/* 	   | 1,0 | 2,0 | 1,1 | 2,1 | 1,2 | 2,2 | 1,3 | 2,3 |  --->  time */
+
+/*   Update linear discriminant function history each frame: */
+    if (*half == 1) {
+	voice[0] = voice[2];
+	voice[1] = voice[3];
+	voice[2] = voice[4];
+	voice[3] = voice[5];
+	*maxmin = *maxamd / max(*minamd,1.f);
+    }
+/*   Calculate voicing parameters twice per frame: */
+    vparms_(&vwin[1], &inbuf[inbuf_offset], &lpbuf[lpbuf_offset], &buflim[1], 
+	    half, dither, mintau, &zc, &lbe, &fbe, &qs, &rc1, &ar_b__, &
+	    ar_f__);
+/*   Estimate signal-to-noise ratio to select the appropriate VDC vector. 
+*/
+/*   The SNR is estimated as the running average of the ratio of the */
+/*   running average full-band voiced energy to the running average */
+/*   full-band unvoiced energy. SNR filter has gain of 63. */
+    r__1 = (*snr + *fbve / (real) max(*fbue,1)) * 63 / 64.f;
+    *snr = (real) i_nint(&r__1);
+    snr2 = *snr * *fbue / max(*lbue,1);
+/*   Quantize SNR to SNRL according to VDCL thresholds. */
+    snrl = 1;
+    i__1 = nvdcl - 1;
+    for (snrl = 1; snrl <= i__1; ++snrl) {
+	if (snr2 > vdcl[snrl - 1]) {
+	    goto L69;
+	}
+    }
+/*   	(Note:  SNRL = NVDCL here) */
+L69:
+/*   Linear discriminant voicing parameters: */
+    value[0] = *maxmin;
+    value[1] = (real) lbe / max(*lbve,1);
+    value[2] = (real) zc;
+    value[3] = rc1;
+    value[4] = qs;
+    value[5] = ivrc[2];
+    value[6] = ar_b__;
+    value[7] = ar_f__;
+/*   Evaluation of linear discriminant function: */
+    voice[*half + 3] = vdc[snrl * 10 - 1];
+    for (i__ = 1; i__ <= 8; ++i__) {
+	voice[*half + 3] += vdc[i__ + snrl * 10 - 11] * value[i__ - 1];
+    }
+/*   Classify as voiced if discriminant > 0, otherwise unvoiced */
+/*   Voicing decision for current half-frame:  1 = Voiced; 0 = Unvoiced */
+    if (voice[*half + 3] > 0.f) {
+	voibuf[*half + 6] = 1;
+    } else {
+	voibuf[*half + 6] = 0;
+    }
+/*   Skip voicing decision smoothing in first half-frame: */
+/*     Give a value to VSTATE, so that trace statements below will print 
+*/
+/*     a consistent value from one call to the next when HALF .EQ. 1. */
+/*     The value of VSTATE is not used for any other purpose when this is 
+*/
+/*     true. */
+    vstate = -1;
+    if (*half == 1) {
+	goto L99;
+    }
+/*   Voicing decision smoothing rules (override of linear combination): */
+
+/* 	Unvoiced half-frames:  At least two in a row. */
+/* 	-------------------- */
+
+/* 	Voiced half-frames:    At least two in a row in one frame. */
+/* 	-------------------    Otherwise at least three in a row. */
+/* 			       (Due to the way transition frames are encoded) */
+
+/* 	In many cases, the discriminant function determines how to smooth. */
+/*	In the following chart, the decisions marked with a * may be overridden
+.*/
+
+/*   Voicing override of transitions at onsets: */
+/* 	If a V/UV or UV/V voicing decision transition occurs within one-half 
+*/
+/* 	frame of an onset bounding a voicing window, then the transition is */
+/* 	moved to occur at the onset. */
+
+/* 	P	1F */
+/* 	-----	----- */
+/* 	0   0   0   0 */
+/* 	0   0   0*  1	(If there is an onset there) */
+/* 	0   0   1*  0*	(Based on 2F and discriminant distance) */
+/* 	0   0   1   1 */
+/* 	0   1*  0   0	(Always) */
+/* 	0   1*  0*  1	(Based on discriminant distance) */
+/* 	0*  1   1   0*	(Based on past, 2F, and discriminant distance) */
+/* 	0   1*  1   1	(If there is an onset there) */
+/* 	1   0*  0   0	(If there is an onset there) */
+/* 	1   0   0   1 */
+/* 	1   0*  1*  0	(Based on discriminant distance) */
+/* 	1   0*  1   1	(Always) */
+/* 	1   1   0   0 */
+/* 	1   1   0*  1*	(Based on 2F and discriminant distance) */
+/* 	1   1   1*  0	(If there is an onset there) */
+/* 	1   1   1   1 */
+
+/*   Determine if there is an onset transition between P and 1F. */
+/*   OT (Onset Transition) is true if there is an onset between */
+/*   P and 1F but not after 1F. */
+    ot = ((obound[1] & 2) != 0 || obound[2] == 1) && (obound[3] & 1) == 0;
+/*   Multi-way dispatch on voicing decision history: */
+    vstate = (voibuf[3] << 3) + (voibuf[4] << 2) + (voibuf[5] << 1) + voibuf[
+	    6];
+    switch (vstate + 1) {
+	case 1:  goto L99;
+	case 2:  goto L1;
+	case 3:  goto L2;
+	case 4:  goto L99;
+	case 5:  goto L4;
+	case 6:  goto L5;
+	case 7:  goto L6;
+	case 8:  goto L7;
+	case 9:  goto L8;
+	case 10:  goto L99;
+	case 11:  goto L10;
+	case 12:  goto L11;
+	case 13:  goto L99;
+	case 14:  goto L13;
+	case 15:  goto L14;
+	case 16:  goto L99;
+    }
+L1:
+    if (ot && voibuf[7] == 1) {
+	voibuf[5] = 1;
+    }
+    goto L99;
+L2:
+    if (voibuf[7] == 0 || voice[2] < -voice[3]) {
+	voibuf[5] = 0;
+    } else {
+	voibuf[6] = 1;
+    }
+    goto L99;
+L4:
+    voibuf[4] = 0;
+    goto L99;
+L5:
+    if (voice[1] < -voice[2]) {
+	voibuf[4] = 0;
+    } else {
+	voibuf[5] = 1;
+    }
+    goto L99;
+/*   VOIBUF(2,0) must be 0 */
+L6:
+    if (voibuf[1] == 1 || voibuf[7] == 1 || voice[3] > voice[0]) {
+	voibuf[6] = 1;
+    } else {
+	voibuf[3] = 1;
+    }
+    goto L99;
+L7:
+    if (ot) {
+	voibuf[4] = 0;
+    }
+    goto L99;
+L8:
+    if (ot) {
+	voibuf[4] = 1;
+    }
+    goto L99;
+L10:
+    if (voice[2] < -voice[1]) {
+	voibuf[5] = 0;
+    } else {
+	voibuf[4] = 1;
+    }
+    goto L99;
+L11:
+    voibuf[4] = 1;
+    goto L99;
+L13:
+    if (voibuf[7] == 0 && voice[3] < -voice[2]) {
+	voibuf[6] = 0;
+    } else {
+	voibuf[5] = 1;
+    }
+    goto L99;
+L14:
+    if (ot && voibuf[7] == 0) {
+	voibuf[5] = 0;
+    }
+/* 	GOTO 99 */
+L99:
+/*   Now update parameters: */
+/*   ---------------------- */
+
+/*  During unvoiced half-frames, update the low band and full band unvoice
+d*/
+/*   energy estimates (LBUE and FBUE) and also the zero crossing */
+/*   threshold (DITHER).  (The input to the unvoiced energy filters is */
+/*   restricted to be less than 10dB above the previous inputs of the */
+/*   filters.) */
+/*   During voiced half-frames, update the low-pass (LBVE) and all-pass */
+/*   (FBVE) voiced energy estimates. */
+    if (voibuf[*half + 6] == 0) {
+/* Computing MIN */
+	i__1 = fbe, i__2 = *ofbue * 3;
+	r__1 = (*sfbue * 63 + (min(i__1,i__2) << 3)) / 64.f;
+	*sfbue = i_nint(&r__1);
+	*fbue = *sfbue / 8;
+	*ofbue = fbe;
+/* Computing MIN */
+	i__1 = lbe, i__2 = *olbue * 3;
+	r__1 = (*slbue * 63 + (min(i__1,i__2) << 3)) / 64.f;
+	*slbue = i_nint(&r__1);
+	*lbue = *slbue / 8;
+	*olbue = lbe;
+    } else {
+	r__1 = (*lbve * 63 + lbe) / 64.f;
+	*lbve = i_nint(&r__1);
+	r__1 = (*fbve * 63 + fbe) / 64.f;
+	*fbve = i_nint(&r__1);
+    }
+/*   Set dither threshold to yield proper zero crossing rates in the */
+/*   presence of low frequency noise and low level signal input. */
+/*   NOTE: The divisor is a function of REF, the expected energies. */
+/* Computing MIN */
+/* Computing MAX */
+    r__2 = sqrt((real) (*lbue * *lbve)) * 64 / 3000;
+    r__1 = max(r__2,1.f);
+    *dither = min(r__1,20.f);
+/*   Voicing decisions are returned in VOIBUF. */
+    return 0;
+} /* voicin_ */

codecs/lpc10/vparms.c

@@ -0,0 +1,243 @@
+/*
+
+$Log$
+Revision 1.4  2000/01/05 08:20:40  markster
+Version 0.1.5 from FTP
+
+Revision 1.2  2000/01/05 08:20:40  markster
+Some OSS fixes and a few lpc changes to make it actually work
+
+ * Revision 1.1  1996/08/19  22:30:04  jaf
+ * Initial revision
+ *
+
+*/
+
+#ifdef P_R_O_T_O_T_Y_P_E_S
+extern int vparms_(integer *vwin, real *inbuf, real *lpbuf, integer *buflim, integer *half, real *dither, integer *mintau, integer *zc, integer *lbe, integer *fbe, real *qs, real *rc1, real *ar_b__, real *ar_f__);
+#endif
+
+/*  -- translated by f2c (version 19951025).
+   You must link the resulting object file with the libraries:
+	-lf2c -lm   (in that order)
+*/
+
+#include "f2c.h"
+
+/* Table of constant values */
+
+static real c_b2 = 1.f;
+
+/* ********************************************************************* */
+
+/* 	VPARMS Version 50 */
+
+/* $Log$
+ * Revision 1.4  2000/01/05 08:20:40  markster
+ * Version 0.1.5 from FTP
+ *
+/* Revision 1.2  2000/01/05 08:20:40  markster
+/* Some OSS fixes and a few lpc changes to make it actually work
+/*
+ * Revision 1.1  1996/08/19  22:30:04  jaf
+ * Initial revision
+ * */
+/* Revision 1.6  1996/03/29  18:01:16  jaf */
+/* Added some more comments about the range of INBUF and LPBUF that can */
+/* be read.  Note that it is possible for index VWIN(2)+1 to be read from */
+/* INBUF, which might be outside of its defined range, although that will */
+/* require more careful checking. */
+
+/* Revision 1.5  1996/03/19  00:02:02  jaf */
+/* I just noticed that the argument DITHER is modified inside of this */
+/* subroutine.  Comments were added explaining the possible final values. */
+
+/* Revision 1.4  1996/03/18  22:22:59  jaf */
+/* Finishing the job I said I did with the last check-in comments. */
+
+/* Revision 1.3  1996/03/18  22:22:17  jaf */
+/* Just added a few comments about which array indices of the arguments */
+/* are used, and mentioning that this subroutine has no local state. */
+
+/* Revision 1.2  1996/03/13  15:02:58  jaf */
+/* Comments added explaining that none of the local variables of this */
+/* subroutine need to be saved from one invocation to the next. */
+
+/* Revision 1.1  1996/02/07 14:50:42  jaf */
+/* Initial revision */
+
+
+/* ********************************************************************* */
+
+/*  Calculate voicing parameters: */
+
+/* Input: */
+/*  VWIN   - Voicing window limits */
+/*           Indices 1 through 2 read. */
+/*  INBUF  - Input speech buffer */
+/*           Indices START-1 through STOP read, */
+/*          where START and STOP are defined in the code (only written once).
+*/
+/*           Note that STOP can be as large as VWIN(2)+1 ! */
+/*  LPBUF  - Low pass filtered speech */
+/*           Indices START-MINTAU through STOP+MINTAU read, */
+/*          where START and STOP are defined in the code (only written once).
+*/
+/*  BUFLIM - Array bounds for INBUF and LPBUF */
+/*           Indices 1 through 4 read. */
+/*  HALF   - Half frame (1 or 2) */
+/*  MINTAU - Lag corresponding to minimum AMDF value (pitch estimate) */
+/* Input/Output: */
+/*  DITHER - Zero crossing threshold */
+/*           The resulting value might be the negation of the input */
+/*           value.  It might always be the same as the input value, */
+/*           if the DO loop below always executes an even number of times. */
+/* Output: (all of them are written on every call) */
+/*  ZC     - Zero crossing rate */
+/*  LBE    - Low band energy (sum of magnitudes - SM) */
+/*  FBE    - Full band energy (SM) */
+/*  QS     - Ratio of 6 dB/oct preemphasized energy to full band energy */
+/*  RC1    - First reflection coefficient */
+/*  AR_B   - Product of the causal forward and reverse pitch */
+/*           prediction gains */
+/*  AR_F   - Product of the noncausal forward and reverse pitch */
+/*           prediction gains */
+/* Internal: */
+/*  OLDSGN - Previous sign of dithered signal */
+/*  VLEN   - Length of voicing window */
+/*  START  - Lower address of current half of voicing window */
+/*  STOP   - Upper address of current half of voicing window */
+/*  E_0    - Energy of LPF speech (sum of squares - SS) */
+/*  E_B    - Energy of LPF speech backward one pitch period (SS) */
+/*  E_F    - Energy of LPF speech forward one pitch period (SS) */
+/*  R_B    - Autocovariance of LPF speech backward one pitch period */
+/*  R_F    - Autocovariance of LPF speech forward one pitch period */
+/*  LP_RMS - Energy of LPF speech (sum of magnitudes - SM) */
+/*  AP_RMS - Energy of all-pass speech (SM) */
+/*  E_PRE  - Energy of 6dB preemphasized speech (SM) */
+/*  E0AP   - Energy of all-pass speech (SS) */
+
+/* This subroutine has no local state. */
+
+/* Subroutine */ int vparms_(integer *vwin, real *inbuf, real *lpbuf, integer 
+	*buflim, integer *half, real *dither, integer *mintau, integer *zc, 
+	integer *lbe, integer *fbe, real *qs, real *rc1, real *ar_b__, real *
+	ar_f__)
+{
+    /* System generated locals */
+    integer inbuf_offset, lpbuf_offset, i__1;
+    real r__1, r__2;
+
+    /* Builtin functions */
+    double r_sign(real *, real *);
+    integer i_nint(real *);
+
+    /* Local variables */
+    integer vlen, stop, i__;
+    real e_pre__;
+    integer start;
+    real ap_rms__, e_0__, oldsgn, lp_rms__, e_b__, e_f__, r_b__, r_f__, e0ap;
+
+/*       Arguments */
+/*       Local variables that need not be saved */
+/*   Calculate zero crossings (ZC) and several energy and correlation */
+/*   measures on low band and full band speech.  Each measure is taken */
+/*   over either the first or the second half of the voicing window, */
+/*   depending on the variable HALF. */
+    /* Parameter adjustments */
+    --vwin;
+    --buflim;
+    lpbuf_offset = buflim[3];
+    lpbuf -= lpbuf_offset;
+    inbuf_offset = buflim[1];
+    inbuf -= inbuf_offset;
+
+    /* Function Body */
+    lp_rms__ = 0.f;
+    ap_rms__ = 0.f;
+    e_pre__ = 0.f;
+    e0ap = 0.f;
+    *rc1 = 0.f;
+    e_0__ = 0.f;
+    e_b__ = 0.f;
+    e_f__ = 0.f;
+    r_f__ = 0.f;
+    r_b__ = 0.f;
+    *zc = 0;
+    vlen = vwin[2] - vwin[1] + 1;
+    start = vwin[1] + (*half - 1) * vlen / 2 + 1;
+    stop = start + vlen / 2 - 1;
+
+/* I'll use the symbol HVL in the table below to represent the value */
+/* VLEN/2.  Note that if VLEN is odd, then HVL should be rounded down, */
+/* i.e., HVL = (VLEN-1)/2. */
+
+/* HALF  START          STOP */
+
+/* 1     VWIN(1)+1      VWIN(1)+HVL */
+/* 2     VWIN(1)+HVL+1  VWIN(1)+2*HVL */
+
+/* Note that if VLEN is even and HALF is 2, then STOP will be */
+/* VWIN(1)+VLEN = VWIN(2)+1.  That could be bad, if that index of INBUF */
+/* is undefined. */
+
+    r__1 = inbuf[start - 1] - *dither;
+    oldsgn = r_sign(&c_b2, &r__1);
+    i__1 = stop;
+    for (i__ = start; i__ <= i__1; ++i__) {
+	lp_rms__ += (r__1 = lpbuf[i__], abs(r__1));
+	ap_rms__ += (r__1 = inbuf[i__], abs(r__1));
+	e_pre__ += (r__1 = inbuf[i__] - inbuf[i__ - 1], abs(r__1));
+/* Computing 2nd power */
+	r__1 = inbuf[i__];
+	e0ap += r__1 * r__1;
+	*rc1 += inbuf[i__] * inbuf[i__ - 1];
+/* Computing 2nd power */
+	r__1 = lpbuf[i__];
+	e_0__ += r__1 * r__1;
+/* Computing 2nd power */
+	r__1 = lpbuf[i__ - *mintau];
+	e_b__ += r__1 * r__1;
+/* Computing 2nd power */
+	r__1 = lpbuf[i__ + *mintau];
+	e_f__ += r__1 * r__1;
+	r_f__ += lpbuf[i__] * lpbuf[i__ + *mintau];
+	r_b__ += lpbuf[i__] * lpbuf[i__ - *mintau];
+	r__1 = inbuf[i__] + *dither;
+	if (r_sign(&c_b2, &r__1) != oldsgn) {
+	    ++(*zc);
+	    oldsgn = -oldsgn;
+	}
+	*dither = -(*dither);
+    }
+/*   Normalized short-term autocovariance coefficient at unit sample delay
+ */
+    *rc1 /= max(e0ap,1.f);
+/*  Ratio of the energy of the first difference signal (6 dB/oct preemphas
+is)*/
+/*   to the energy of the full band signal */
+/* Computing MAX */
+    r__1 = ap_rms__ * 2.f;
+    *qs = e_pre__ / max(r__1,1.f);
+/*   aR_b is the product of the forward and reverse prediction gains, */
+/*   looking backward in time (the causal case). */
+    *ar_b__ = r_b__ / max(e_b__,1.f) * (r_b__ / max(e_0__,1.f));
+/*  aR_f is the same as aR_b, but looking forward in time (non causal case
+).*/
+    *ar_f__ = r_f__ / max(e_f__,1.f) * (r_f__ / max(e_0__,1.f));
+/*   Normalize ZC, LBE, and FBE to old fixed window length of 180. */
+/*   (The fraction 90/VLEN has a range of .58 to 1) */
+    r__2 = (real) (*zc << 1);
+    r__1 = r__2 * (90.f / vlen);
+    *zc = i_nint(&r__1);
+/* Computing MIN */
+    r__1 = lp_rms__ / 4 * (90.f / vlen);
+    i__1 = i_nint(&r__1);
+    *lbe = min(i__1,32767);
+/* Computing MIN */
+    r__1 = ap_rms__ / 4 * (90.f / vlen);
+    i__1 = i_nint(&r__1);
+    *fbe = min(i__1,32767);
+    return 0;
+} /* vparms_ */
+

codecs/lpc10_slin_ex.h

@@ -0,0 +1,13 @@
+/*
+  * 8-bit raw data
+  *
+  * Source: example.lpc10
+  *
+  * Copyright (C) 1999, Mark Spencer and Linux Support Services
+  *
+  * Distributed under the terms of the GNU General Public License
+  *
+  */
+
+static unsigned char lpc10_slin_ex[] = {
+0x1, 0x8, 0x31, 0x8, 0x31, 0x80, 0x30 };

codecs/mp3/Makefile

@@ -0,0 +1,37 @@
+#
+# LMC section
+
+CFLAGS+= -I../include -Iinclude -O6 -funroll-loops -finline-functions -Wall -Wno-missing-prototypes -Wno-missing-declarations -g
+RANLIB=ranlib
+
+# the XING decoder objs and dependencies:
+# This is kinda nasty, since there's C, C++, and asm, oh my!
+# of course, each needs different compilation methods. grr.
+XINGOBJX86 = src/x86gas.o
+
+XINGOBJS = src/cdct.o src/cupl3.o \
+  src/hwin.o src/iup.o src/l3init.o \
+  src/msis.o src/wavep.o src/csbt.o \
+  src/cwinm.o src/icdct.o src/mdct.o \
+  src/uph.o src/cup.o src/dec8.o \
+  src/isbt.o src/l3dq.o src/mhead.o \
+  src/upsf.o src/iwinm.o
+
+LIBMP3=libmp3.a
+ARFLAGS=cr
+
+XINGLMCOBJC += $(shell if uname -m | grep -q i.86; then echo src/x86gas.o; fi)
+
+#assembly lang code, if we need it
+
+XINGLMCOBJ = $(XINGOBJS)
+
+all: $(LIBMP3)
+
+$(LIBMP3): $(XINGOBJS)
+	$(AR) $(ARFLAGS) $(LIBMP3) $(XINGLMCOBJ)
+	$(RANLIB) $(LIBMP3)
+
+clean:
+	rm -f $(XINGOBJS)
+	rm -f $(LIBMP3)

codecs/mp3/include/L3.h

@@ -0,0 +1,177 @@
+/*____________________________________________________________________________
+	
+	FreeAmp - The Free MP3 Player
+
+        MP3 Decoder originally Copyright (C) 1996-1997 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.
+	
+	$Id$
+____________________________________________________________________________*/
+
+/****  L3.h  ***************************************************
+
+  Layer III structures
+
+  *** Layer III is 32 bit only          ***
+  *** Layer III code assumes 32 bit int ***
+
+******************************************************************/
+
+#define GLOBAL_GAIN_SCALE (4*15)
+/* #define GLOBAL_GAIN_SCALE 0 */
+
+
+#ifdef _M_IX86
+#define LITTLE_ENDIAN 1
+#endif
+
+#ifdef _M_ALPHA
+#define LITTLE_ENDIAN 1
+#endif
+
+#ifdef sparc
+#define LITTLE_ENDIAN 0
+#endif
+
+#ifndef LITTLE_ENDIAN
+#error Layer III LITTLE_ENDIAN must be defined 0 or 1
+#endif
+
+/*-----------------------------------------------------------*/
+/*---- huffman lookup tables ---*/
+/* endian dependent !!! */
+#if LITTLE_ENDIAN
+typedef union
+{
+   int ptr;
+   struct
+   {
+      unsigned char signbits;
+      unsigned char x;
+      unsigned char y;
+      unsigned char purgebits;	// 0 = esc
+
+   }
+   b;
+}
+HUFF_ELEMENT;
+
+#else /* big endian machines */
+typedef union
+{
+   int ptr;			/* int must be 32 bits or more */
+   struct
+   {
+      unsigned char purgebits;	// 0 = esc
+
+      unsigned char y;
+      unsigned char x;
+      unsigned char signbits;
+   }
+   b;
+}
+HUFF_ELEMENT;
+
+#endif
+/*--------------------------------------------------------------*/
+typedef struct
+{
+   unsigned int bitbuf;
+   int bits;
+   unsigned char *bs_ptr;
+   unsigned char *bs_ptr0;
+   unsigned char *bs_ptr_end;	// optional for overrun test
+
+}
+BITDAT;
+
+/*-- side info ---*/
+typedef struct
+{
+   int part2_3_length;
+   int big_values;
+   int global_gain;
+   int scalefac_compress;
+   int window_switching_flag;
+   int block_type;
+   int mixed_block_flag;
+   int table_select[3];
+   int subblock_gain[3];
+   int region0_count;
+   int region1_count;
+   int preflag;
+   int scalefac_scale;
+   int count1table_select;
+}
+GR;
+typedef struct
+{
+   int mode;
+   int mode_ext;
+/*---------------*/
+   int main_data_begin;		/* beginning, not end, my spec wrong */
+   int private_bits;
+/*---------------*/
+   int scfsi[2];		/* 4 bit flags [ch] */
+   GR gr[2][2];			/* [gran][ch] */
+}
+SIDE_INFO;
+
+/*-----------------------------------------------------------*/
+/*-- scale factors ---*/
+// check dimensions - need 21 long, 3*12 short
+// plus extra for implicit sf=0 above highest cb
+typedef struct
+{
+   int l[23];			/* [cb] */
+   int s[3][13];		/* [window][cb] */
+}
+SCALEFACT;
+
+/*-----------------------------------------------------------*/
+typedef struct
+{
+   int cbtype;			/* long=0 short=1 */
+   int cbmax;			/* max crit band */
+//   int lb_type;			/* long block type 0 1 3 */
+   int cbs0;			/* short band start index 0 3 12 (12=no shorts */
+   int ncbl;			/* number long cb's 0 8 21 */
+   int cbmax_s[3];		/* cbmax by individual short blocks */
+}
+CB_INFO;
+
+/*-----------------------------------------------------------*/
+/* scale factor infor for MPEG2 intensity stereo  */
+typedef struct
+{
+   int nr[3];
+   int slen[3];
+   int intensity_scale;
+}
+IS_SF_INFO;
+
+/*-----------------------------------------------------------*/
+typedef union
+{
+   int s;
+   float x;
+}
+SAMPLE;
+
+/*-----------------------------------------------------------*/

codecs/mp3/include/htable.h

@@ -0,0 +1,999 @@
+/*____________________________________________________________________________
+	
+	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.
+	
+	$Id$
+____________________________________________________________________________*/
+
+/* TABLE  1    4 entries  maxbits  3  linbits  0 */
+static HUFF_ELEMENT huff_table_1[] =
+{
+  {0xFF000003}, {0x03010102}, {0x03010001}, {0x02000101}, {0x02000101},	/*  4 */
+  {0x01000000}, {0x01000000}, {0x01000000}, {0x01000000},
+};
+
+/* max table bits  3 */
+
+/* TABLE  2    9 entries  maxbits  6  linbits  0 */
+static HUFF_ELEMENT huff_table_2[] =
+{
+  {0xFF000006}, {0x06020202}, {0x06020001}, {0x05020102}, {0x05020102},	/*  4 */
+  {0x05010202}, {0x05010202}, {0x05000201}, {0x05000201}, {0x03010102},	/*  9 */
+  {0x03010102}, {0x03010102}, {0x03010102}, {0x03010102}, {0x03010102},	/* 14 */
+  {0x03010102}, {0x03010102}, {0x03010001}, {0x03010001}, {0x03010001},	/* 19 */
+  {0x03010001}, {0x03010001}, {0x03010001}, {0x03010001}, {0x03010001},	/* 24 */
+  {0x03000101}, {0x03000101}, {0x03000101}, {0x03000101}, {0x03000101},	/* 29 */
+  {0x03000101}, {0x03000101}, {0x03000101}, {0x01000000}, {0x01000000},	/* 34 */
+  {0x01000000}, {0x01000000}, {0x01000000}, {0x01000000}, {0x01000000},	/* 39 */
+  {0x01000000}, {0x01000000}, {0x01000000}, {0x01000000}, {0x01000000},	/* 44 */
+  {0x01000000}, {0x01000000}, {0x01000000}, {0x01000000}, {0x01000000},	/* 49 */
+  {0x01000000}, {0x01000000}, {0x01000000}, {0x01000000}, {0x01000000},	/* 54 */
+  {0x01000000}, {0x01000000}, {0x01000000}, {0x01000000}, {0x01000000},	/* 59 */
+  {0x01000000}, {0x01000000}, {0x01000000}, {0x01000000}, {0x01000000}, /* 64 */
+};
+
+/* max table bits  6 */
+
+/* TABLE  3    9 entries  maxbits  6  linbits  0 */
+static HUFF_ELEMENT huff_table_3[] =
+{
+  {0xFF000006}, {0x06020202}, {0x06020001}, {0x05020102}, {0x05020102},	/*  4 */
+  {0x05010202}, {0x05010202}, {0x05000201}, {0x05000201}, {0x03000101},	/*  9 */
+  {0x03000101}, {0x03000101}, {0x03000101}, {0x03000101}, {0x03000101},	/* 14 */
+  {0x03000101}, {0x03000101}, {0x02010102}, {0x02010102}, {0x02010102},	/* 19 */
+  {0x02010102}, {0x02010102}, {0x02010102}, {0x02010102}, {0x02010102},	/* 24 */
+  {0x02010102}, {0x02010102}, {0x02010102}, {0x02010102}, {0x02010102},	/* 29 */
+  {0x02010102}, {0x02010102}, {0x02010102}, {0x02010001}, {0x02010001},	/* 34 */
+  {0x02010001}, {0x02010001}, {0x02010001}, {0x02010001}, {0x02010001},	/* 39 */
+  {0x02010001}, {0x02010001}, {0x02010001}, {0x02010001}, {0x02010001},	/* 44 */
+  {0x02010001}, {0x02010001}, {0x02010001}, {0x02010001}, {0x02000000},	/* 49 */
+  {0x02000000}, {0x02000000}, {0x02000000}, {0x02000000}, {0x02000000},	/* 54 */
+  {0x02000000}, {0x02000000}, {0x02000000}, {0x02000000}, {0x02000000},	/* 59 */
+  {0x02000000}, {0x02000000}, {0x02000000}, {0x02000000}, {0x02000000}, /* 64 */
+};
+
+/* max table bits  6 */
+/* NO XING TABLE 4 */
+
+/* TABLE  5   16 entries  maxbits  8  linbits  0 */
+static HUFF_ELEMENT huff_table_5[] =
+{
+  {0xFF000008}, {0x08030302}, {0x08030202}, {0x07020302}, {0x07020302},	/*  4 */
+  {0x06010302}, {0x06010302}, {0x06010302}, {0x06010302}, {0x07030102},	/*  9 */
+  {0x07030102}, {0x07030001}, {0x07030001}, {0x07000301}, {0x07000301},	/* 14 */
+  {0x07020202}, {0x07020202}, {0x06020102}, {0x06020102}, {0x06020102},	/* 19 */
+  {0x06020102}, {0x06010202}, {0x06010202}, {0x06010202}, {0x06010202},	/* 24 */
+  {0x06020001}, {0x06020001}, {0x06020001}, {0x06020001}, {0x06000201},	/* 29 */
+  {0x06000201}, {0x06000201}, {0x06000201}, {0x03010102}, {0x03010102},	/* 34 */
+  {0x03010102}, {0x03010102}, {0x03010102}, {0x03010102}, {0x03010102},	/* 39 */
+  {0x03010102}, {0x03010102}, {0x03010102}, {0x03010102}, {0x03010102},	/* 44 */
+  {0x03010102}, {0x03010102}, {0x03010102}, {0x03010102}, {0x03010102},	/* 49 */
+  {0x03010102}, {0x03010102}, {0x03010102}, {0x03010102}, {0x03010102},	/* 54 */
+  {0x03010102}, {0x03010102}, {0x03010102}, {0x03010102}, {0x03010102},	/* 59 */
+  {0x03010102}, {0x03010102}, {0x03010102}, {0x03010102}, {0x03010102},	/* 64 */
+  {0x03010001}, {0x03010001}, {0x03010001}, {0x03010001}, {0x03010001},	/* 69 */
+  {0x03010001}, {0x03010001}, {0x03010001}, {0x03010001}, {0x03010001},	/* 74 */
+  {0x03010001}, {0x03010001}, {0x03010001}, {0x03010001}, {0x03010001},	/* 79 */
+  {0x03010001}, {0x03010001}, {0x03010001}, {0x03010001}, {0x03010001},	/* 84 */
+  {0x03010001}, {0x03010001}, {0x03010001}, {0x03010001}, {0x03010001},	/* 89 */
+  {0x03010001}, {0x03010001}, {0x03010001}, {0x03010001}, {0x03010001},	/* 94 */
+  {0x03010001}, {0x03010001}, {0x03000101}, {0x03000101}, {0x03000101},	/* 99 */
+  {0x03000101}, {0x03000101}, {0x03000101}, {0x03000101}, {0x03000101},	/* 104 */
+  {0x03000101}, {0x03000101}, {0x03000101}, {0x03000101}, {0x03000101},	/* 109 */
+  {0x03000101}, {0x03000101}, {0x03000101}, {0x03000101}, {0x03000101},	/* 114 */
+  {0x03000101}, {0x03000101}, {0x03000101}, {0x03000101}, {0x03000101},	/* 119 */
+  {0x03000101}, {0x03000101}, {0x03000101}, {0x03000101}, {0x03000101},	/* 124 */
+  {0x03000101}, {0x03000101}, {0x03000101}, {0x03000101}, {0x01000000},	/* 129 */
+  {0x01000000}, {0x01000000}, {0x01000000}, {0x01000000}, {0x01000000},	/* 134 */
+  {0x01000000}, {0x01000000}, {0x01000000}, {0x01000000}, {0x01000000},	/* 139 */
+  {0x01000000}, {0x01000000}, {0x01000000}, {0x01000000}, {0x01000000},	/* 144 */
+  {0x01000000}, {0x01000000}, {0x01000000}, {0x01000000}, {0x01000000},	/* 149 */
+  {0x01000000}, {0x01000000}, {0x01000000}, {0x01000000}, {0x01000000},	/* 154 */
+  {0x01000000}, {0x01000000}, {0x01000000}, {0x01000000}, {0x01000000},	/* 159 */
+  {0x01000000}, {0x01000000}, {0x01000000}, {0x01000000}, {0x01000000},	/* 164 */
+  {0x01000000}, {0x01000000}, {0x01000000}, {0x01000000}, {0x01000000},	/* 169 */
+  {0x01000000}, {0x01000000}, {0x01000000}, {0x01000000}, {0x01000000},	/* 174 */
+  {0x01000000}, {0x01000000}, {0x01000000}, {0x01000000}, {0x01000000},	/* 179 */
+  {0x01000000}, {0x01000000}, {0x01000000}, {0x01000000}, {0x01000000},	/* 184 */
+  {0x01000000}, {0x01000000}, {0x01000000}, {0x01000000}, {0x01000000},	/* 189 */
+  {0x01000000}, {0x01000000}, {0x01000000}, {0x01000000}, {0x01000000},	/* 194 */
+  {0x01000000}, {0x01000000}, {0x01000000}, {0x01000000}, {0x01000000},	/* 199 */
+  {0x01000000}, {0x01000000}, {0x01000000}, {0x01000000}, {0x01000000},	/* 204 */
+  {0x01000000}, {0x01000000}, {0x01000000}, {0x01000000}, {0x01000000},	/* 209 */
+  {0x01000000}, {0x01000000}, {0x01000000}, {0x01000000}, {0x01000000},	/* 214 */
+  {0x01000000}, {0x01000000}, {0x01000000}, {0x01000000}, {0x01000000},	/* 219 */
+  {0x01000000}, {0x01000000}, {0x01000000}, {0x01000000}, {0x01000000},	/* 224 */
+  {0x01000000}, {0x01000000}, {0x01000000}, {0x01000000}, {0x01000000},	/* 229 */
+  {0x01000000}, {0x01000000}, {0x01000000}, {0x01000000}, {0x01000000},	/* 234 */
+  {0x01000000}, {0x01000000}, {0x01000000}, {0x01000000}, {0x01000000},	/* 239 */
+  {0x01000000}, {0x01000000}, {0x01000000}, {0x01000000}, {0x01000000},	/* 244 */
+  {0x01000000}, {0x01000000}, {0x01000000}, {0x01000000}, {0x01000000},	/* 249 */
+  {0x01000000}, {0x01000000}, {0x01000000}, {0x01000000}, {0x01000000},	/* 254 */
+  {0x01000000}, {0x01000000},
+};
+
+/* max table bits  8 */
+
+/* TABLE  6   16 entries  maxbits  7  linbits  0 */
+static HUFF_ELEMENT huff_table_6[] =
+{
+  {0xFF000007}, {0x07030302}, {0x07030001}, {0x06030202}, {0x06030202},	/*  4 */
+  {0x06020302}, {0x06020302}, {0x06000301}, {0x06000301}, {0x05030102},	/*  9 */
+  {0x05030102}, {0x05030102}, {0x05030102}, {0x05010302}, {0x05010302},	/* 14 */
+  {0x05010302}, {0x05010302}, {0x05020202}, {0x05020202}, {0x05020202},	/* 19 */
+  {0x05020202}, {0x05020001}, {0x05020001}, {0x05020001}, {0x05020001},	/* 24 */
+  {0x04020102}, {0x04020102}, {0x04020102}, {0x04020102}, {0x04020102},	/* 29 */
+  {0x04020102}, {0x04020102}, {0x04020102}, {0x04010202}, {0x04010202},	/* 34 */
+  {0x04010202}, {0x04010202}, {0x04010202}, {0x04010202}, {0x04010202},	/* 39 */
+  {0x04010202}, {0x04000201}, {0x04000201}, {0x04000201}, {0x04000201},	/* 44 */
+  {0x04000201}, {0x04000201}, {0x04000201}, {0x04000201}, {0x03010001},	/* 49 */
+  {0x03010001}, {0x03010001}, {0x03010001}, {0x03010001}, {0x03010001},	/* 54 */
+  {0x03010001}, {0x03010001}, {0x03010001}, {0x03010001}, {0x03010001},	/* 59 */
+  {0x03010001}, {0x03010001}, {0x03010001}, {0x03010001}, {0x03010001},	/* 64 */
+  {0x02010102}, {0x02010102}, {0x02010102}, {0x02010102}, {0x02010102},	/* 69 */
+  {0x02010102}, {0x02010102}, {0x02010102}, {0x02010102}, {0x02010102},	/* 74 */
+  {0x02010102}, {0x02010102}, {0x02010102}, {0x02010102}, {0x02010102},	/* 79 */
+  {0x02010102}, {0x02010102}, {0x02010102}, {0x02010102}, {0x02010102},	/* 84 */
+  {0x02010102}, {0x02010102}, {0x02010102}, {0x02010102}, {0x02010102},	/* 89 */
+  {0x02010102}, {0x02010102}, {0x02010102}, {0x02010102}, {0x02010102},	/* 94 */
+  {0x02010102}, {0x02010102}, {0x03000101}, {0x03000101}, {0x03000101},	/* 99 */
+  {0x03000101}, {0x03000101}, {0x03000101}, {0x03000101}, {0x03000101},	/* 104 */
+  {0x03000101}, {0x03000101}, {0x03000101}, {0x03000101}, {0x03000101},	/* 109 */
+  {0x03000101}, {0x03000101}, {0x03000101}, {0x03000000}, {0x03000000},	/* 114 */
+  {0x03000000}, {0x03000000}, {0x03000000}, {0x03000000}, {0x03000000},	/* 119 */
+  {0x03000000}, {0x03000000}, {0x03000000}, {0x03000000}, {0x03000000},	/* 124 */
+  {0x03000000}, {0x03000000}, {0x03000000}, {0x03000000},
+};
+
+/* max table bits  7 */
+
+/* TABLE  7   36 entries  maxbits 10  linbits  0 */
+static HUFF_ELEMENT huff_table_7[] =
+{
+  {0xFF000006}, {0x00000041}, {0x00000052}, {0x0000005B}, {0x00000060},	/*  4 */
+  {0x00000063}, {0x00000068}, {0x0000006B}, {0x06020102}, {0x05010202},	/*  9 */
+  {0x05010202}, {0x06020001}, {0x06000201}, {0x04010102}, {0x04010102},	/* 14 */
+  {0x04010102}, {0x04010102}, {0x03010001}, {0x03010001}, {0x03010001},	/* 19 */
+  {0x03010001}, {0x03010001}, {0x03010001}, {0x03010001}, {0x03010001},	/* 24 */
+  {0x03000101}, {0x03000101}, {0x03000101}, {0x03000101}, {0x03000101},	/* 29 */
+  {0x03000101}, {0x03000101}, {0x03000101}, {0x01000000}, {0x01000000},	/* 34 */
+  {0x01000000}, {0x01000000}, {0x01000000}, {0x01000000}, {0x01000000},	/* 39 */
+  {0x01000000}, {0x01000000}, {0x01000000}, {0x01000000}, {0x01000000},	/* 44 */
+  {0x01000000}, {0x01000000}, {0x01000000}, {0x01000000}, {0x01000000},	/* 49 */
+  {0x01000000}, {0x01000000}, {0x01000000}, {0x01000000}, {0x01000000},	/* 54 */
+  {0x01000000}, {0x01000000}, {0x01000000}, {0x01000000}, {0x01000000},	/* 59 */
+  {0x01000000}, {0x01000000}, {0x01000000}, {0x01000000}, {0x01000000},	/* 64 */
+  {0xFF000004}, {0x04050502}, {0x04050402}, {0x04040502}, {0x04030502},	/* 69 */
+  {0x03050302}, {0x03050302}, {0x03040402}, {0x03040402}, {0x03050202},	/* 74 */
+  {0x03050202}, {0x03020502}, {0x03020502}, {0x02050102}, {0x02050102},	/* 79 */
+  {0x02050102}, {0x02050102}, {0xFF000003}, {0x02010502}, {0x02010502},	/* 84 */
+  {0x03050001}, {0x03040302}, {0x02000501}, {0x02000501}, {0x03030402},	/* 89 */
+  {0x03030302}, {0xFF000002}, {0x02040202}, {0x02020402}, {0x01040102},	/* 94 */
+  {0x01040102}, {0xFF000001}, {0x01010402}, {0x01000401}, {0xFF000002},	/* 99 */
+  {0x02040001}, {0x02030202}, {0x02020302}, {0x02030001}, {0xFF000001},	/* 104 */
+  {0x01030102}, {0x01010302}, {0xFF000001}, {0x01000301}, {0x01020202}, /* 109 */
+};
+
+/* max table bits  6 */
+
+/* TABLE  8   36 entries  maxbits 11  linbits  0 */
+static HUFF_ELEMENT huff_table_8[] =
+{
+  {0xFF000008}, {0x00000101}, {0x0000010A}, {0x0000010F}, {0x08050102},	/*  4 */
+  {0x08010502}, {0x00000112}, {0x00000115}, {0x08040202}, {0x08020402},	/*  9 */
+  {0x08040102}, {0x07010402}, {0x07010402}, {0x08040001}, {0x08000401},	/* 14 */
+  {0x08030202}, {0x08020302}, {0x08030102}, {0x08010302}, {0x08030001},	/* 19 */
+  {0x08000301}, {0x06020202}, {0x06020202}, {0x06020202}, {0x06020202},	/* 24 */
+  {0x06020001}, {0x06020001}, {0x06020001}, {0x06020001}, {0x06000201},	/* 29 */
+  {0x06000201}, {0x06000201}, {0x06000201}, {0x04020102}, {0x04020102},	/* 34 */
+  {0x04020102}, {0x04020102}, {0x04020102}, {0x04020102}, {0x04020102},	/* 39 */
+  {0x04020102}, {0x04020102}, {0x04020102}, {0x04020102}, {0x04020102},	/* 44 */
+  {0x04020102}, {0x04020102}, {0x04020102}, {0x04020102}, {0x04010202},	/* 49 */
+  {0x04010202}, {0x04010202}, {0x04010202}, {0x04010202}, {0x04010202},	/* 54 */
+  {0x04010202}, {0x04010202}, {0x04010202}, {0x04010202}, {0x04010202},	/* 59 */
+  {0x04010202}, {0x04010202}, {0x04010202}, {0x04010202}, {0x04010202},	/* 64 */
+  {0x02010102}, {0x02010102}, {0x02010102}, {0x02010102}, {0x02010102},	/* 69 */
+  {0x02010102}, {0x02010102}, {0x02010102}, {0x02010102}, {0x02010102},	/* 74 */
+  {0x02010102}, {0x02010102}, {0x02010102}, {0x02010102}, {0x02010102},	/* 79 */
+  {0x02010102}, {0x02010102}, {0x02010102}, {0x02010102}, {0x02010102},	/* 84 */
+  {0x02010102}, {0x02010102}, {0x02010102}, {0x02010102}, {0x02010102},	/* 89 */
+  {0x02010102}, {0x02010102}, {0x02010102}, {0x02010102}, {0x02010102},	/* 94 */
+  {0x02010102}, {0x02010102}, {0x02010102}, {0x02010102}, {0x02010102},	/* 99 */
+  {0x02010102}, {0x02010102}, {0x02010102}, {0x02010102}, {0x02010102},	/* 104 */
+  {0x02010102}, {0x02010102}, {0x02010102}, {0x02010102}, {0x02010102},	/* 109 */
+  {0x02010102}, {0x02010102}, {0x02010102}, {0x02010102}, {0x02010102},	/* 114 */
+  {0x02010102}, {0x02010102}, {0x02010102}, {0x02010102}, {0x02010102},	/* 119 */
+  {0x02010102}, {0x02010102}, {0x02010102}, {0x02010102}, {0x02010102},	/* 124 */
+  {0x02010102}, {0x02010102}, {0x02010102}, {0x02010102}, {0x03010001},	/* 129 */
+  {0x03010001}, {0x03010001}, {0x03010001}, {0x03010001}, {0x03010001},	/* 134 */
+  {0x03010001}, {0x03010001}, {0x03010001}, {0x03010001}, {0x03010001},	/* 139 */
+  {0x03010001}, {0x03010001}, {0x03010001}, {0x03010001}, {0x03010001},	/* 144 */
+  {0x03010001}, {0x03010001}, {0x03010001}, {0x03010001}, {0x03010001},	/* 149 */
+  {0x03010001}, {0x03010001}, {0x03010001}, {0x03010001}, {0x03010001},	/* 154 */
+  {0x03010001}, {0x03010001}, {0x03010001}, {0x03010001}, {0x03010001},	/* 159 */
+  {0x03010001}, {0x03000101}, {0x03000101}, {0x03000101}, {0x03000101},	/* 164 */
+  {0x03000101}, {0x03000101}, {0x03000101}, {0x03000101}, {0x03000101},	/* 169 */
+  {0x03000101}, {0x03000101}, {0x03000101}, {0x03000101}, {0x03000101},	/* 174 */
+  {0x03000101}, {0x03000101}, {0x03000101}, {0x03000101}, {0x03000101},	/* 179 */
+  {0x03000101}, {0x03000101}, {0x03000101}, {0x03000101}, {0x03000101},	/* 184 */
+  {0x03000101}, {0x03000101}, {0x03000101}, {0x03000101}, {0x03000101},	/* 189 */
+  {0x03000101}, {0x03000101}, {0x03000101}, {0x02000000}, {0x02000000},	/* 194 */
+  {0x02000000}, {0x02000000}, {0x02000000}, {0x02000000}, {0x02000000},	/* 199 */
+  {0x02000000}, {0x02000000}, {0x02000000}, {0x02000000}, {0x02000000},	/* 204 */
+  {0x02000000}, {0x02000000}, {0x02000000}, {0x02000000}, {0x02000000},	/* 209 */
+  {0x02000000}, {0x02000000}, {0x02000000}, {0x02000000}, {0x02000000},	/* 214 */
+  {0x02000000}, {0x02000000}, {0x02000000}, {0x02000000}, {0x02000000},	/* 219 */
+  {0x02000000}, {0x02000000}, {0x02000000}, {0x02000000}, {0x02000000},	/* 224 */
+  {0x02000000}, {0x02000000}, {0x02000000}, {0x02000000}, {0x02000000},	/* 229 */
+  {0x02000000}, {0x02000000}, {0x02000000}, {0x02000000}, {0x02000000},	/* 234 */
+  {0x02000000}, {0x02000000}, {0x02000000}, {0x02000000}, {0x02000000},	/* 239 */
+  {0x02000000}, {0x02000000}, {0x02000000}, {0x02000000}, {0x02000000},	/* 244 */
+  {0x02000000}, {0x02000000}, {0x02000000}, {0x02000000}, {0x02000000},	/* 249 */
+  {0x02000000}, {0x02000000}, {0x02000000}, {0x02000000}, {0x02000000},	/* 254 */
+  {0x02000000}, {0x02000000}, {0xFF000003}, {0x03050502}, {0x03040502},	/* 259 */
+  {0x02050402}, {0x02050402}, {0x01030502}, {0x01030502}, {0x01030502},	/* 264 */
+  {0x01030502}, {0xFF000002}, {0x02050302}, {0x02040402}, {0x01050202},	/* 269 */
+  {0x01050202}, {0xFF000001}, {0x01020502}, {0x01050001}, {0xFF000001},	/* 274 */
+  {0x01040302}, {0x01030402}, {0xFF000001}, {0x01000501}, {0x01030302}, /* 279 */
+};
+
+/* max table bits  8 */
+
+/* TABLE  9   36 entries  maxbits  9  linbits  0 */
+static HUFF_ELEMENT huff_table_9[] =
+{
+  {0xFF000006}, {0x00000041}, {0x0000004A}, {0x0000004F}, {0x00000052},	/*  4 */
+  {0x00000057}, {0x0000005A}, {0x06040102}, {0x06010402}, {0x06030202},	/*  9 */
+  {0x06020302}, {0x05030102}, {0x05030102}, {0x05010302}, {0x05010302},	/* 14 */
+  {0x06030001}, {0x06000301}, {0x05020202}, {0x05020202}, {0x05020001},	/* 19 */
+  {0x05020001}, {0x04020102}, {0x04020102}, {0x04020102}, {0x04020102},	/* 24 */
+  {0x04010202}, {0x04010202}, {0x04010202}, {0x04010202}, {0x04000201},	/* 29 */
+  {0x04000201}, {0x04000201}, {0x04000201}, {0x03010102}, {0x03010102},	/* 34 */
+  {0x03010102}, {0x03010102}, {0x03010102}, {0x03010102}, {0x03010102},	/* 39 */
+  {0x03010102}, {0x03010001}, {0x03010001}, {0x03010001}, {0x03010001},	/* 44 */
+  {0x03010001}, {0x03010001}, {0x03010001}, {0x03010001}, {0x03000101},	/* 49 */
+  {0x03000101}, {0x03000101}, {0x03000101}, {0x03000101}, {0x03000101},	/* 54 */
+  {0x03000101}, {0x03000101}, {0x03000000}, {0x03000000}, {0x03000000},	/* 59 */
+  {0x03000000}, {0x03000000}, {0x03000000}, {0x03000000}, {0x03000000},	/* 64 */
+  {0xFF000003}, {0x03050502}, {0x03050402}, {0x02050302}, {0x02050302},	/* 69 */
+  {0x02030502}, {0x02030502}, {0x03040502}, {0x03050001}, {0xFF000002},	/* 74 */
+  {0x02040402}, {0x02050202}, {0x02020502}, {0x02050102}, {0xFF000001},	/* 79 */
+  {0x01010502}, {0x01040302}, {0xFF000002}, {0x01030402}, {0x01030402},	/* 84 */
+  {0x02000501}, {0x02040001}, {0xFF000001}, {0x01040202}, {0x01020402},	/* 89 */
+  {0xFF000001}, {0x01030302}, {0x01000401},
+};
+
+/* max table bits  6 */
+
+/* TABLE 10   64 entries  maxbits 11  linbits  0 */
+static HUFF_ELEMENT huff_table_10[] =
+{
+  {0xFF000008}, {0x00000101}, {0x0000010A}, {0x0000010F}, {0x00000118},	/*  4 */
+  {0x0000011B}, {0x00000120}, {0x00000125}, {0x08070102}, {0x08010702},	/*  9 */
+  {0x0000012A}, {0x0000012D}, {0x00000132}, {0x08060102}, {0x08010602},	/* 14 */
+  {0x08000601}, {0x00000137}, {0x0000013A}, {0x0000013D}, {0x08040102},	/* 19 */
+  {0x08010402}, {0x08000401}, {0x08030202}, {0x08020302}, {0x08030001},	/* 24 */
+  {0x07030102}, {0x07030102}, {0x07010302}, {0x07010302}, {0x07000301},	/* 29 */
+  {0x07000301}, {0x07020202}, {0x07020202}, {0x06020102}, {0x06020102},	/* 34 */
+  {0x06020102}, {0x06020102}, {0x06010202}, {0x06010202}, {0x06010202},	/* 39 */
+  {0x06010202}, {0x06020001}, {0x06020001}, {0x06020001}, {0x06020001},	/* 44 */
+  {0x06000201}, {0x06000201}, {0x06000201}, {0x06000201}, {0x04010102},	/* 49 */
+  {0x04010102}, {0x04010102}, {0x04010102}, {0x04010102}, {0x04010102},	/* 54 */
+  {0x04010102}, {0x04010102}, {0x04010102}, {0x04010102}, {0x04010102},	/* 59 */
+  {0x04010102}, {0x04010102}, {0x04010102}, {0x04010102}, {0x04010102},	/* 64 */
+  {0x03010001}, {0x03010001}, {0x03010001}, {0x03010001}, {0x03010001},	/* 69 */
+  {0x03010001}, {0x03010001}, {0x03010001}, {0x03010001}, {0x03010001},	/* 74 */
+  {0x03010001}, {0x03010001}, {0x03010001}, {0x03010001}, {0x03010001},	/* 79 */
+  {0x03010001}, {0x03010001}, {0x03010001}, {0x03010001}, {0x03010001},	/* 84 */
+  {0x03010001}, {0x03010001}, {0x03010001}, {0x03010001}, {0x03010001},	/* 89 */
+  {0x03010001}, {0x03010001}, {0x03010001}, {0x03010001}, {0x03010001},	/* 94 */
+  {0x03010001}, {0x03010001}, {0x03000101}, {0x03000101}, {0x03000101},	/* 99 */
+  {0x03000101}, {0x03000101}, {0x03000101}, {0x03000101}, {0x03000101},	/* 104 */
+  {0x03000101}, {0x03000101}, {0x03000101}, {0x03000101}, {0x03000101},	/* 109 */
+  {0x03000101}, {0x03000101}, {0x03000101}, {0x03000101}, {0x03000101},	/* 114 */
+  {0x03000101}, {0x03000101}, {0x03000101}, {0x03000101}, {0x03000101},	/* 119 */
+  {0x03000101}, {0x03000101}, {0x03000101}, {0x03000101}, {0x03000101},	/* 124 */
+  {0x03000101}, {0x03000101}, {0x03000101}, {0x03000101}, {0x01000000},	/* 129 */
+  {0x01000000}, {0x01000000}, {0x01000000}, {0x01000000}, {0x01000000},	/* 134 */
+  {0x01000000}, {0x01000000}, {0x01000000}, {0x01000000}, {0x01000000},	/* 139 */
+  {0x01000000}, {0x01000000}, {0x01000000}, {0x01000000}, {0x01000000},	/* 144 */
+  {0x01000000}, {0x01000000}, {0x01000000}, {0x01000000}, {0x01000000},	/* 149 */
+  {0x01000000}, {0x01000000}, {0x01000000}, {0x01000000}, {0x01000000},	/* 154 */
+  {0x01000000}, {0x01000000}, {0x01000000}, {0x01000000}, {0x01000000},	/* 159 */
+  {0x01000000}, {0x01000000}, {0x01000000}, {0x01000000}, {0x01000000},	/* 164 */
+  {0x01000000}, {0x01000000}, {0x01000000}, {0x01000000}, {0x01000000},	/* 169 */
+  {0x01000000}, {0x01000000}, {0x01000000}, {0x01000000}, {0x01000000},	/* 174 */
+  {0x01000000}, {0x01000000}, {0x01000000}, {0x01000000}, {0x01000000},	/* 179 */
+  {0x01000000}, {0x01000000}, {0x01000000}, {0x01000000}, {0x01000000},	/* 184 */
+  {0x01000000}, {0x01000000}, {0x01000000}, {0x01000000}, {0x01000000},	/* 189 */
+  {0x01000000}, {0x01000000}, {0x01000000}, {0x01000000}, {0x01000000},	/* 194 */
+  {0x01000000}, {0x01000000}, {0x01000000}, {0x01000000}, {0x01000000},	/* 199 */
+  {0x01000000}, {0x01000000}, {0x01000000}, {0x01000000}, {0x01000000},	/* 204 */
+  {0x01000000}, {0x01000000}, {0x01000000}, {0x01000000}, {0x01000000},	/* 209 */
+  {0x01000000}, {0x01000000}, {0x01000000}, {0x01000000}, {0x01000000},	/* 214 */
+  {0x01000000}, {0x01000000}, {0x01000000}, {0x01000000}, {0x01000000},	/* 219 */
+  {0x01000000}, {0x01000000}, {0x01000000}, {0x01000000}, {0x01000000},	/* 224 */
+  {0x01000000}, {0x01000000}, {0x01000000}, {0x01000000}, {0x01000000},	/* 229 */
+  {0x01000000}, {0x01000000}, {0x01000000}, {0x01000000}, {0x01000000},	/* 234 */
+  {0x01000000}, {0x01000000}, {0x01000000}, {0x01000000}, {0x01000000},	/* 239 */
+  {0x01000000}, {0x01000000}, {0x01000000}, {0x01000000}, {0x01000000},	/* 244 */
+  {0x01000000}, {0x01000000}, {0x01000000}, {0x01000000}, {0x01000000},	/* 249 */
+  {0x01000000}, {0x01000000}, {0x01000000}, {0x01000000}, {0x01000000},	/* 254 */
+  {0x01000000}, {0x01000000}, {0xFF000003}, {0x03070702}, {0x03070602},	/* 259 */
+  {0x03060702}, {0x03070502}, {0x03050702}, {0x03060602}, {0x02070402},	/* 264 */
+  {0x02070402}, {0xFF000002}, {0x02040702}, {0x02060502}, {0x02050602},	/* 269 */
+  {0x02070302}, {0xFF000003}, {0x02030702}, {0x02030702}, {0x02060402},	/* 274 */
+  {0x02060402}, {0x03050502}, {0x03040502}, {0x02030602}, {0x02030602},	/* 279 */
+  {0xFF000001}, {0x01070202}, {0x01020702}, {0xFF000002}, {0x02040602},	/* 284 */
+  {0x02070001}, {0x01000701}, {0x01000701}, {0xFF000002}, {0x01020602},	/* 289 */
+  {0x01020602}, {0x02050402}, {0x02050302}, {0xFF000002}, {0x01060001},	/* 294 */
+  {0x01060001}, {0x02030502}, {0x02040402}, {0xFF000001}, {0x01060302},	/* 299 */
+  {0x01060202}, {0xFF000002}, {0x02050202}, {0x02020502}, {0x01050102},	/* 304 */
+  {0x01050102}, {0xFF000002}, {0x01010502}, {0x01010502}, {0x02040302},	/* 309 */
+  {0x02030402}, {0xFF000001}, {0x01050001}, {0x01000501}, {0xFF000001},	/* 314 */
+  {0x01040202}, {0x01020402}, {0xFF000001}, {0x01030302}, {0x01040001}, /* 319 */
+};
+
+/* max table bits  8 */
+
+/* TABLE 11   64 entries  maxbits 11  linbits  0 */
+static HUFF_ELEMENT huff_table_11[] =
+{
+  {0xFF000008}, {0x00000101}, {0x00000106}, {0x0000010F}, {0x00000114},	/*  4 */
+  {0x00000117}, {0x08070202}, {0x08020702}, {0x0000011C}, {0x07010702},	/*  9 */
+  {0x07010702}, {0x08070102}, {0x08000701}, {0x08060302}, {0x08030602},	/* 14 */
+  {0x08000601}, {0x0000011F}, {0x00000122}, {0x08050102}, {0x07020602},	/* 19 */
+  {0x07020602}, {0x08060202}, {0x08060001}, {0x07060102}, {0x07060102},	/* 24 */
+  {0x07010602}, {0x07010602}, {0x08010502}, {0x08040302}, {0x08000501},	/* 29 */
+  {0x00000125}, {0x08040202}, {0x08020402}, {0x08040102}, {0x08010402},	/* 34 */
+  {0x08040001}, {0x08000401}, {0x07030202}, {0x07030202}, {0x07020302},	/* 39 */
+  {0x07020302}, {0x06030102}, {0x06030102}, {0x06030102}, {0x06030102},	/* 44 */
+  {0x06010302}, {0x06010302}, {0x06010302}, {0x06010302}, {0x07030001},	/* 49 */
+  {0x07030001}, {0x07000301}, {0x07000301}, {0x06020202}, {0x06020202},	/* 54 */
+  {0x06020202}, {0x06020202}, {0x05010202}, {0x05010202}, {0x05010202},	/* 59 */
+  {0x05010202}, {0x05010202}, {0x05010202}, {0x05010202}, {0x05010202},	/* 64 */
+  {0x04020102}, {0x04020102}, {0x04020102}, {0x04020102}, {0x04020102},	/* 69 */
+  {0x04020102}, {0x04020102}, {0x04020102}, {0x04020102}, {0x04020102},	/* 74 */
+  {0x04020102}, {0x04020102}, {0x04020102}, {0x04020102}, {0x04020102},	/* 79 */
+  {0x04020102}, {0x05020001}, {0x05020001}, {0x05020001}, {0x05020001},	/* 84 */
+  {0x05020001}, {0x05020001}, {0x05020001}, {0x05020001}, {0x05000201},	/* 89 */
+  {0x05000201}, {0x05000201}, {0x05000201}, {0x05000201}, {0x05000201},	/* 94 */
+  {0x05000201}, {0x05000201}, {0x03010102}, {0x03010102}, {0x03010102},	/* 99 */
+  {0x03010102}, {0x03010102}, {0x03010102}, {0x03010102}, {0x03010102},	/* 104 */
+  {0x03010102}, {0x03010102}, {0x03010102}, {0x03010102}, {0x03010102},	/* 109 */
+  {0x03010102}, {0x03010102}, {0x03010102}, {0x03010102}, {0x03010102},	/* 114 */
+  {0x03010102}, {0x03010102}, {0x03010102}, {0x03010102}, {0x03010102},	/* 119 */
+  {0x03010102}, {0x03010102}, {0x03010102}, {0x03010102}, {0x03010102},	/* 124 */
+  {0x03010102}, {0x03010102}, {0x03010102}, {0x03010102}, {0x03010001},	/* 129 */
+  {0x03010001}, {0x03010001}, {0x03010001}, {0x03010001}, {0x03010001},	/* 134 */
+  {0x03010001}, {0x03010001}, {0x03010001}, {0x03010001}, {0x03010001},	/* 139 */
+  {0x03010001}, {0x03010001}, {0x03010001}, {0x03010001}, {0x03010001},	/* 144 */
+  {0x03010001}, {0x03010001}, {0x03010001}, {0x03010001}, {0x03010001},	/* 149 */
+  {0x03010001}, {0x03010001}, {0x03010001}, {0x03010001}, {0x03010001},	/* 154 */
+  {0x03010001}, {0x03010001}, {0x03010001}, {0x03010001}, {0x03010001},	/* 159 */
+  {0x03010001}, {0x03000101}, {0x03000101}, {0x03000101}, {0x03000101},	/* 164 */
+  {0x03000101}, {0x03000101}, {0x03000101}, {0x03000101}, {0x03000101},	/* 169 */
+  {0x03000101}, {0x03000101}, {0x03000101}, {0x03000101}, {0x03000101},	/* 174 */
+  {0x03000101}, {0x03000101}, {0x03000101}, {0x03000101}, {0x03000101},	/* 179 */
+  {0x03000101}, {0x03000101}, {0x03000101}, {0x03000101}, {0x03000101},	/* 184 */
+  {0x03000101}, {0x03000101}, {0x03000101}, {0x03000101}, {0x03000101},	/* 189 */
+  {0x03000101}, {0x03000101}, {0x03000101}, {0x02000000}, {0x02000000},	/* 194 */
+  {0x02000000}, {0x02000000}, {0x02000000}, {0x02000000}, {0x02000000},	/* 199 */
+  {0x02000000}, {0x02000000}, {0x02000000}, {0x02000000}, {0x02000000},	/* 204 */
+  {0x02000000}, {0x02000000}, {0x02000000}, {0x02000000}, {0x02000000},	/* 209 */
+  {0x02000000}, {0x02000000}, {0x02000000}, {0x02000000}, {0x02000000},	/* 214 */
+  {0x02000000}, {0x02000000}, {0x02000000}, {0x02000000}, {0x02000000},	/* 219 */
+  {0x02000000}, {0x02000000}, {0x02000000}, {0x02000000}, {0x02000000},	/* 224 */
+  {0x02000000}, {0x02000000}, {0x02000000}, {0x02000000}, {0x02000000},	/* 229 */
+  {0x02000000}, {0x02000000}, {0x02000000}, {0x02000000}, {0x02000000},	/* 234 */
+  {0x02000000}, {0x02000000}, {0x02000000}, {0x02000000}, {0x02000000},	/* 239 */
+  {0x02000000}, {0x02000000}, {0x02000000}, {0x02000000}, {0x02000000},	/* 244 */
+  {0x02000000}, {0x02000000}, {0x02000000}, {0x02000000}, {0x02000000},	/* 249 */
+  {0x02000000}, {0x02000000}, {0x02000000}, {0x02000000}, {0x02000000},	/* 254 */
+  {0x02000000}, {0x02000000}, {0xFF000002}, {0x02070702}, {0x02070602},	/* 259 */
+  {0x02060702}, {0x02050702}, {0xFF000003}, {0x02060602}, {0x02060602},	/* 264 */
+  {0x02070402}, {0x02070402}, {0x02040702}, {0x02040702}, {0x03070502},	/* 269 */
+  {0x03050502}, {0xFF000002}, {0x02060502}, {0x02050602}, {0x01070302},	/* 274 */
+  {0x01070302}, {0xFF000001}, {0x01030702}, {0x01060402}, {0xFF000002},	/* 279 */
+  {0x02050402}, {0x02040502}, {0x02050302}, {0x02030502}, {0xFF000001},	/* 284 */
+  {0x01040602}, {0x01070001}, {0xFF000001}, {0x01040402}, {0x01050202},	/* 289 */
+  {0xFF000001}, {0x01020502}, {0x01050001}, {0xFF000001}, {0x01030402},	/* 294 */
+  {0x01030302},
+};
+
+/* max table bits  8 */
+
+/* TABLE 12   64 entries  maxbits 10  linbits  0 */
+static HUFF_ELEMENT huff_table_12[] =
+{
+  {0xFF000007}, {0x00000081}, {0x0000008A}, {0x0000008F}, {0x00000092},	/*  4 */
+  {0x00000097}, {0x0000009A}, {0x0000009D}, {0x000000A2}, {0x000000A5},	/*  9 */
+  {0x000000A8}, {0x07060202}, {0x07020602}, {0x07010602}, {0x000000AD},	/* 14 */
+  {0x000000B0}, {0x000000B3}, {0x07050102}, {0x07010502}, {0x07040302},	/* 19 */
+  {0x07030402}, {0x000000B6}, {0x07040202}, {0x07020402}, {0x07040102},	/* 24 */
+  {0x06030302}, {0x06030302}, {0x06010402}, {0x06010402}, {0x06030202},	/* 29 */
+  {0x06030202}, {0x06020302}, {0x06020302}, {0x07000401}, {0x07030001},	/* 34 */
+  {0x06000301}, {0x06000301}, {0x05030102}, {0x05030102}, {0x05030102},	/* 39 */
+  {0x05030102}, {0x05010302}, {0x05010302}, {0x05010302}, {0x05010302},	/* 44 */
+  {0x05020202}, {0x05020202}, {0x05020202}, {0x05020202}, {0x04020102},	/* 49 */
+  {0x04020102}, {0x04020102}, {0x04020102}, {0x04020102}, {0x04020102},	/* 54 */
+  {0x04020102}, {0x04020102}, {0x04010202}, {0x04010202}, {0x04010202},	/* 59 */
+  {0x04010202}, {0x04010202}, {0x04010202}, {0x04010202}, {0x04010202},	/* 64 */
+  {0x05020001}, {0x05020001}, {0x05020001}, {0x05020001}, {0x05000201},	/* 69 */
+  {0x05000201}, {0x05000201}, {0x05000201}, {0x04000000}, {0x04000000},	/* 74 */
+  {0x04000000}, {0x04000000}, {0x04000000}, {0x04000000}, {0x04000000},	/* 79 */
+  {0x04000000}, {0x03010102}, {0x03010102}, {0x03010102}, {0x03010102},	/* 84 */
+  {0x03010102}, {0x03010102}, {0x03010102}, {0x03010102}, {0x03010102},	/* 89 */
+  {0x03010102}, {0x03010102}, {0x03010102}, {0x03010102}, {0x03010102},	/* 94 */
+  {0x03010102}, {0x03010102}, {0x03010001}, {0x03010001}, {0x03010001},	/* 99 */
+  {0x03010001}, {0x03010001}, {0x03010001}, {0x03010001}, {0x03010001},	/* 104 */
+  {0x03010001}, {0x03010001}, {0x03010001}, {0x03010001}, {0x03010001},	/* 109 */
+  {0x03010001}, {0x03010001}, {0x03010001}, {0x03000101}, {0x03000101},	/* 114 */
+  {0x03000101}, {0x03000101}, {0x03000101}, {0x03000101}, {0x03000101},	/* 119 */
+  {0x03000101}, {0x03000101}, {0x03000101}, {0x03000101}, {0x03000101},	/* 124 */
+  {0x03000101}, {0x03000101}, {0x03000101}, {0x03000101}, {0xFF000003},	/* 129 */
+  {0x03070702}, {0x03070602}, {0x02060702}, {0x02060702}, {0x02070502},	/* 134 */
+  {0x02070502}, {0x02050702}, {0x02050702}, {0xFF000002}, {0x02060602},	/* 139 */
+  {0x02070402}, {0x02040702}, {0x02050602}, {0xFF000001}, {0x01060502},	/* 144 */
+  {0x01070302}, {0xFF000002}, {0x02030702}, {0x02050502}, {0x01070202},	/* 149 */
+  {0x01070202}, {0xFF000001}, {0x01020702}, {0x01060402}, {0xFF000001},	/* 154 */
+  {0x01040602}, {0x01070102}, {0xFF000002}, {0x01010702}, {0x01010702},	/* 159 */
+  {0x02070001}, {0x02000701}, {0xFF000001}, {0x01060302}, {0x01030602},	/* 164 */
+  {0xFF000001}, {0x01050402}, {0x01040502}, {0xFF000002}, {0x01040402},	/* 169 */
+  {0x01040402}, {0x02060001}, {0x02050001}, {0xFF000001}, {0x01060102},	/* 174 */
+  {0x01000601}, {0xFF000001}, {0x01050302}, {0x01030502}, {0xFF000001},	/* 179 */
+  {0x01050202}, {0x01020502}, {0xFF000001}, {0x01000501}, {0x01040001}, /* 184 */
+};
+
+/* max table bits  7 */
+
+/* TABLE 13  256 entries  maxbits 19  linbits  0 */
+static HUFF_ELEMENT huff_table_13[] =
+{
+  {0xFF000006}, {0x00000041}, {0x00000082}, {0x000000C3}, {0x000000E4},	/*  4 */
+  {0x00000105}, {0x00000116}, {0x0000011F}, {0x00000130}, {0x00000139},	/*  9 */
+  {0x0000013E}, {0x00000143}, {0x00000146}, {0x06020102}, {0x06010202},	/* 14 */
+  {0x06020001}, {0x06000201}, {0x04010102}, {0x04010102}, {0x04010102},	/* 19 */
+  {0x04010102}, {0x04010001}, {0x04010001}, {0x04010001}, {0x04010001},	/* 24 */
+  {0x03000101}, {0x03000101}, {0x03000101}, {0x03000101}, {0x03000101},	/* 29 */
+  {0x03000101}, {0x03000101}, {0x03000101}, {0x01000000}, {0x01000000},	/* 34 */
+  {0x01000000}, {0x01000000}, {0x01000000}, {0x01000000}, {0x01000000},	/* 39 */
+  {0x01000000}, {0x01000000}, {0x01000000}, {0x01000000}, {0x01000000},	/* 44 */
+  {0x01000000}, {0x01000000}, {0x01000000}, {0x01000000}, {0x01000000},	/* 49 */
+  {0x01000000}, {0x01000000}, {0x01000000}, {0x01000000}, {0x01000000},	/* 54 */
+  {0x01000000}, {0x01000000}, {0x01000000}, {0x01000000}, {0x01000000},	/* 59 */
+  {0x01000000}, {0x01000000}, {0x01000000}, {0x01000000}, {0x01000000},	/* 64 */
+  {0xFF000006}, {0x00000108}, {0x00000111}, {0x0000011A}, {0x00000123},	/* 69 */
+  {0x0000012C}, {0x00000131}, {0x00000136}, {0x0000013F}, {0x00000144},	/* 74 */
+  {0x00000147}, {0x0000014C}, {0x00000151}, {0x00000156}, {0x0000015B},	/* 79 */
+  {0x060F0102}, {0x06010F02}, {0x06000F01}, {0x00000160}, {0x00000163},	/* 84 */
+  {0x00000166}, {0x06020E02}, {0x00000169}, {0x060E0102}, {0x06010E02},	/* 89 */
+  {0x0000016C}, {0x0000016F}, {0x00000172}, {0x00000175}, {0x00000178},	/* 94 */
+  {0x0000017B}, {0x06060C02}, {0x060D0302}, {0x0000017E}, {0x060D0202},	/* 99 */
+  {0x06020D02}, {0x060D0102}, {0x06070B02}, {0x00000181}, {0x00000184},	/* 104 */
+  {0x06030C02}, {0x00000187}, {0x060B0402}, {0x05010D02}, {0x05010D02},	/* 109 */
+  {0x060D0001}, {0x06000D01}, {0x060A0802}, {0x06080A02}, {0x060C0402},	/* 114 */
+  {0x06040C02}, {0x060B0602}, {0x06060B02}, {0x050C0302}, {0x050C0302},	/* 119 */
+  {0x050C0202}, {0x050C0202}, {0x05020C02}, {0x05020C02}, {0x050B0502},	/* 124 */
+  {0x050B0502}, {0x06050B02}, {0x06090802}, {0x050C0102}, {0x050C0102},	/* 129 */
+  {0xFF000006}, {0x05010C02}, {0x05010C02}, {0x06080902}, {0x060C0001},	/* 134 */
+  {0x05000C01}, {0x05000C01}, {0x06040B02}, {0x060A0602}, {0x06060A02},	/* 139 */
+  {0x06090702}, {0x050B0302}, {0x050B0302}, {0x05030B02}, {0x05030B02},	/* 144 */
+  {0x06080802}, {0x060A0502}, {0x050B0202}, {0x050B0202}, {0x06050A02},	/* 149 */
+  {0x06090602}, {0x05040A02}, {0x05040A02}, {0x06080702}, {0x06070802},	/* 154 */
+  {0x05040902}, {0x05040902}, {0x06070702}, {0x06060702}, {0x04020B02},	/* 159 */
+  {0x04020B02}, {0x04020B02}, {0x04020B02}, {0x040B0102}, {0x040B0102},	/* 164 */
+  {0x040B0102}, {0x040B0102}, {0x04010B02}, {0x04010B02}, {0x04010B02},	/* 169 */
+  {0x04010B02}, {0x050B0001}, {0x050B0001}, {0x05000B01}, {0x05000B01},	/* 174 */
+  {0x05060902}, {0x05060902}, {0x050A0402}, {0x050A0402}, {0x050A0302},	/* 179 */
+  {0x050A0302}, {0x05030A02}, {0x05030A02}, {0x05090502}, {0x05090502},	/* 184 */
+  {0x05050902}, {0x05050902}, {0x040A0202}, {0x040A0202}, {0x040A0202},	/* 189 */
+  {0x040A0202}, {0x04020A02}, {0x04020A02}, {0x04020A02}, {0x04020A02},	/* 194 */
+  {0xFF000005}, {0x040A0102}, {0x040A0102}, {0x04010A02}, {0x04010A02},	/* 199 */
+  {0x050A0001}, {0x05080602}, {0x04000A01}, {0x04000A01}, {0x05060802},	/* 204 */
+  {0x05090402}, {0x04030902}, {0x04030902}, {0x05090302}, {0x05080502},	/* 209 */
+  {0x05050802}, {0x05070602}, {0x04090202}, {0x04090202}, {0x04020902},	/* 214 */
+  {0x04020902}, {0x05070502}, {0x05050702}, {0x04080302}, {0x04080302},	/* 219 */
+  {0x04030802}, {0x04030802}, {0x05060602}, {0x05070402}, {0x05040702},	/* 224 */
+  {0x05060502}, {0x05050602}, {0x05030702}, {0xFF000005}, {0x03090102},	/* 229 */
+  {0x03090102}, {0x03090102}, {0x03090102}, {0x03010902}, {0x03010902},	/* 234 */
+  {0x03010902}, {0x03010902}, {0x04090001}, {0x04090001}, {0x04000901},	/* 239 */
+  {0x04000901}, {0x04080402}, {0x04080402}, {0x04040802}, {0x04040802},	/* 244 */
+  {0x04020702}, {0x04020702}, {0x05060402}, {0x05040602}, {0x03080202},	/* 249 */
+  {0x03080202}, {0x03080202}, {0x03080202}, {0x03020802}, {0x03020802},	/* 254 */
+  {0x03020802}, {0x03020802}, {0x03080102}, {0x03080102}, {0x03080102},	/* 259 */
+  {0x03080102}, {0xFF000004}, {0x04070302}, {0x04070202}, {0x03070102},	/* 264 */
+  {0x03070102}, {0x03010702}, {0x03010702}, {0x04050502}, {0x04070001},	/* 269 */
+  {0x04000701}, {0x04060302}, {0x04030602}, {0x04050402}, {0x04040502},	/* 274 */
+  {0x04060202}, {0x04020602}, {0x04050302}, {0xFF000003}, {0x02010802},	/* 279 */
+  {0x02010802}, {0x03080001}, {0x03000801}, {0x03060102}, {0x03010602},	/* 284 */
+  {0x03060001}, {0x03000601}, {0xFF000004}, {0x04030502}, {0x04040402},	/* 289 */
+  {0x03050202}, {0x03050202}, {0x03020502}, {0x03020502}, {0x03050001},	/* 294 */
+  {0x03050001}, {0x02050102}, {0x02050102}, {0x02050102}, {0x02050102},	/* 299 */
+  {0x02010502}, {0x02010502}, {0x02010502}, {0x02010502}, {0xFF000003},	/* 304 */
+  {0x03040302}, {0x03030402}, {0x03000501}, {0x03040202}, {0x03020402},	/* 309 */
+  {0x03030302}, {0x02040102}, {0x02040102}, {0xFF000002}, {0x01010402},	/* 314 */
+  {0x01010402}, {0x02040001}, {0x02000401}, {0xFF000002}, {0x02030202},	/* 319 */
+  {0x02020302}, {0x01030102}, {0x01030102}, {0xFF000001}, {0x01010302},	/* 324 */
+  {0x01030001}, {0xFF000001}, {0x01000301}, {0x01020202}, {0xFF000003},	/* 329 */
+  {0x00000082}, {0x0000008B}, {0x0000008E}, {0x00000091}, {0x00000094},	/* 334 */
+  {0x00000097}, {0x030C0E02}, {0x030D0D02}, {0xFF000003}, {0x00000093},	/* 339 */
+  {0x030E0B02}, {0x030B0E02}, {0x030F0902}, {0x03090F02}, {0x030A0E02},	/* 344 */
+  {0x030D0B02}, {0x030B0D02}, {0xFF000003}, {0x030F0802}, {0x03080F02},	/* 349 */
+  {0x030C0C02}, {0x0000008D}, {0x030E0802}, {0x00000090}, {0x02070F02},	/* 354 */
+  {0x02070F02}, {0xFF000003}, {0x020A0D02}, {0x020A0D02}, {0x030D0A02},	/* 359 */
+  {0x030C0B02}, {0x030B0C02}, {0x03060F02}, {0x020F0602}, {0x020F0602},	/* 364 */
+  {0xFF000002}, {0x02080E02}, {0x020F0502}, {0x020D0902}, {0x02090D02},	/* 369 */
+  {0xFF000002}, {0x02050F02}, {0x02070E02}, {0x020C0A02}, {0x020B0B02},	/* 374 */
+  {0xFF000003}, {0x020F0402}, {0x020F0402}, {0x02040F02}, {0x02040F02},	/* 379 */
+  {0x030A0C02}, {0x03060E02}, {0x02030F02}, {0x02030F02}, {0xFF000002},	/* 384 */
+  {0x010F0302}, {0x010F0302}, {0x020D0802}, {0x02080D02}, {0xFF000001},	/* 389 */
+  {0x010F0202}, {0x01020F02}, {0xFF000002}, {0x020E0602}, {0x020C0902},	/* 394 */
+  {0x010F0001}, {0x010F0001}, {0xFF000002}, {0x02090C02}, {0x020E0502},	/* 399 */
+  {0x010B0A02}, {0x010B0A02}, {0xFF000002}, {0x020D0702}, {0x02070D02},	/* 404 */
+  {0x010E0402}, {0x010E0402}, {0xFF000002}, {0x02080C02}, {0x02060D02},	/* 409 */
+  {0x010E0302}, {0x010E0302}, {0xFF000002}, {0x01090B02}, {0x01090B02},	/* 414 */
+  {0x020B0902}, {0x020A0A02}, {0xFF000001}, {0x010A0B02}, {0x01050E02},	/* 419 */
+  {0xFF000001}, {0x01040E02}, {0x010C0802}, {0xFF000001}, {0x010D0602},	/* 424 */
+  {0x01030E02}, {0xFF000001}, {0x010E0202}, {0x010E0001}, {0xFF000001},	/* 429 */
+  {0x01000E01}, {0x010D0502}, {0xFF000001}, {0x01050D02}, {0x010C0702},	/* 434 */
+  {0xFF000001}, {0x01070C02}, {0x010D0402}, {0xFF000001}, {0x010B0802},	/* 439 */
+  {0x01080B02}, {0xFF000001}, {0x01040D02}, {0x010A0902}, {0xFF000001},	/* 444 */
+  {0x01090A02}, {0x010C0602}, {0xFF000001}, {0x01030D02}, {0x010B0702},	/* 449 */
+  {0xFF000001}, {0x010C0502}, {0x01050C02}, {0xFF000001}, {0x01090902},	/* 454 */
+  {0x010A0702}, {0xFF000001}, {0x01070A02}, {0x01070902}, {0xFF000003},	/* 459 */
+  {0x00000023}, {0x030D0F02}, {0x020D0E02}, {0x020D0E02}, {0x010F0F02},	/* 464 */
+  {0x010F0F02}, {0x010F0F02}, {0x010F0F02}, {0xFF000001}, {0x010F0E02},	/* 469 */
+  {0x010F0D02}, {0xFF000001}, {0x010E0E02}, {0x010F0C02}, {0xFF000001},	/* 474 */
+  {0x010E0D02}, {0x010F0B02}, {0xFF000001}, {0x010B0F02}, {0x010E0C02},	/* 479 */
+  {0xFF000002}, {0x010C0D02}, {0x010C0D02}, {0x020F0A02}, {0x02090E02},	/* 484 */
+  {0xFF000001}, {0x010A0F02}, {0x010D0C02}, {0xFF000001}, {0x010E0A02},	/* 489 */
+  {0x010E0902}, {0xFF000001}, {0x010F0702}, {0x010E0702}, {0xFF000001},	/* 494 */
+  {0x010E0F02}, {0x010C0F02},
+};
+
+/* max table bits  6 */
+/* NO XING TABLE 14 */
+
+/* TABLE 15  256 entries  maxbits 13  linbits  0 */
+static HUFF_ELEMENT huff_table_15[] =
+{
+  {0xFF000008}, {0x00000101}, {0x00000122}, {0x00000143}, {0x00000154},	/*  4 */
+  {0x00000165}, {0x00000176}, {0x0000017F}, {0x00000188}, {0x00000199},	/*  9 */
+  {0x000001A2}, {0x000001AB}, {0x000001B4}, {0x000001BD}, {0x000001C2},	/* 14 */
+  {0x000001CB}, {0x000001D4}, {0x000001D9}, {0x000001DE}, {0x000001E3},	/* 19 */
+  {0x000001E8}, {0x000001ED}, {0x000001F2}, {0x000001F7}, {0x000001FC},	/* 24 */
+  {0x00000201}, {0x00000204}, {0x00000207}, {0x0000020A}, {0x0000020F},	/* 29 */
+  {0x00000212}, {0x00000215}, {0x0000021A}, {0x0000021D}, {0x00000220},	/* 34 */
+  {0x08010902}, {0x00000223}, {0x00000226}, {0x00000229}, {0x0000022C},	/* 39 */
+  {0x0000022F}, {0x08080202}, {0x08020802}, {0x08080102}, {0x08010802},	/* 44 */
+  {0x00000232}, {0x00000235}, {0x00000238}, {0x0000023B}, {0x08070202},	/* 49 */
+  {0x08020702}, {0x08040602}, {0x08070102}, {0x08050502}, {0x08010702},	/* 54 */
+  {0x0000023E}, {0x08060302}, {0x08030602}, {0x08050402}, {0x08040502},	/* 59 */
+  {0x08060202}, {0x08020602}, {0x08060102}, {0x00000241}, {0x08050302},	/* 64 */
+  {0x07010602}, {0x07010602}, {0x08030502}, {0x08040402}, {0x07050202},	/* 69 */
+  {0x07050202}, {0x07020502}, {0x07020502}, {0x07050102}, {0x07050102},	/* 74 */
+  {0x07010502}, {0x07010502}, {0x08050001}, {0x08000501}, {0x07040302},	/* 79 */
+  {0x07040302}, {0x07030402}, {0x07030402}, {0x07040202}, {0x07040202},	/* 84 */
+  {0x07020402}, {0x07020402}, {0x07030302}, {0x07030302}, {0x06010402},	/* 89 */
+  {0x06010402}, {0x06010402}, {0x06010402}, {0x07040102}, {0x07040102},	/* 94 */
+  {0x07040001}, {0x07040001}, {0x06030202}, {0x06030202}, {0x06030202},	/* 99 */
+  {0x06030202}, {0x06020302}, {0x06020302}, {0x06020302}, {0x06020302},	/* 104 */
+  {0x07000401}, {0x07000401}, {0x07030001}, {0x07030001}, {0x06030102},	/* 109 */
+  {0x06030102}, {0x06030102}, {0x06030102}, {0x06010302}, {0x06010302},	/* 114 */
+  {0x06010302}, {0x06010302}, {0x06000301}, {0x06000301}, {0x06000301},	/* 119 */
+  {0x06000301}, {0x05020202}, {0x05020202}, {0x05020202}, {0x05020202},	/* 124 */
+  {0x05020202}, {0x05020202}, {0x05020202}, {0x05020202}, {0x05020102},	/* 129 */
+  {0x05020102}, {0x05020102}, {0x05020102}, {0x05020102}, {0x05020102},	/* 134 */
+  {0x05020102}, {0x05020102}, {0x05010202}, {0x05010202}, {0x05010202},	/* 139 */
+  {0x05010202}, {0x05010202}, {0x05010202}, {0x05010202}, {0x05010202},	/* 144 */
+  {0x05020001}, {0x05020001}, {0x05020001}, {0x05020001}, {0x05020001},	/* 149 */
+  {0x05020001}, {0x05020001}, {0x05020001}, {0x05000201}, {0x05000201},	/* 154 */
+  {0x05000201}, {0x05000201}, {0x05000201}, {0x05000201}, {0x05000201},	/* 159 */
+  {0x05000201}, {0x03010102}, {0x03010102}, {0x03010102}, {0x03010102},	/* 164 */
+  {0x03010102}, {0x03010102}, {0x03010102}, {0x03010102}, {0x03010102},	/* 169 */
+  {0x03010102}, {0x03010102}, {0x03010102}, {0x03010102}, {0x03010102},	/* 174 */
+  {0x03010102}, {0x03010102}, {0x03010102}, {0x03010102}, {0x03010102},	/* 179 */
+  {0x03010102}, {0x03010102}, {0x03010102}, {0x03010102}, {0x03010102},	/* 184 */
+  {0x03010102}, {0x03010102}, {0x03010102}, {0x03010102}, {0x03010102},	/* 189 */
+  {0x03010102}, {0x03010102}, {0x03010102}, {0x04010001}, {0x04010001},	/* 194 */
+  {0x04010001}, {0x04010001}, {0x04010001}, {0x04010001}, {0x04010001},	/* 199 */
+  {0x04010001}, {0x04010001}, {0x04010001}, {0x04010001}, {0x04010001},	/* 204 */
+  {0x04010001}, {0x04010001}, {0x04010001}, {0x04010001}, {0x04000101},	/* 209 */
+  {0x04000101}, {0x04000101}, {0x04000101}, {0x04000101}, {0x04000101},	/* 214 */
+  {0x04000101}, {0x04000101}, {0x04000101}, {0x04000101}, {0x04000101},	/* 219 */
+  {0x04000101}, {0x04000101}, {0x04000101}, {0x04000101}, {0x04000101},	/* 224 */
+  {0x03000000}, {0x03000000}, {0x03000000}, {0x03000000}, {0x03000000},	/* 229 */
+  {0x03000000}, {0x03000000}, {0x03000000}, {0x03000000}, {0x03000000},	/* 234 */
+  {0x03000000}, {0x03000000}, {0x03000000}, {0x03000000}, {0x03000000},	/* 239 */
+  {0x03000000}, {0x03000000}, {0x03000000}, {0x03000000}, {0x03000000},	/* 244 */
+  {0x03000000}, {0x03000000}, {0x03000000}, {0x03000000}, {0x03000000},	/* 249 */
+  {0x03000000}, {0x03000000}, {0x03000000}, {0x03000000}, {0x03000000},	/* 254 */
+  {0x03000000}, {0x03000000}, {0xFF000005}, {0x050F0F02}, {0x050F0E02},	/* 259 */
+  {0x050E0F02}, {0x050F0D02}, {0x040E0E02}, {0x040E0E02}, {0x050D0F02},	/* 264 */
+  {0x050F0C02}, {0x050C0F02}, {0x050E0D02}, {0x050D0E02}, {0x050F0B02},	/* 269 */
+  {0x040B0F02}, {0x040B0F02}, {0x050E0C02}, {0x050C0E02}, {0x040D0D02},	/* 274 */
+  {0x040D0D02}, {0x040F0A02}, {0x040F0A02}, {0x040A0F02}, {0x040A0F02},	/* 279 */
+  {0x040E0B02}, {0x040E0B02}, {0x040B0E02}, {0x040B0E02}, {0x040D0C02},	/* 284 */
+  {0x040D0C02}, {0x040C0D02}, {0x040C0D02}, {0x040F0902}, {0x040F0902},	/* 289 */
+  {0xFF000005}, {0x04090F02}, {0x04090F02}, {0x040A0E02}, {0x040A0E02},	/* 294 */
+  {0x040D0B02}, {0x040D0B02}, {0x040B0D02}, {0x040B0D02}, {0x040F0802},	/* 299 */
+  {0x040F0802}, {0x04080F02}, {0x04080F02}, {0x040C0C02}, {0x040C0C02},	/* 304 */
+  {0x040E0902}, {0x040E0902}, {0x04090E02}, {0x04090E02}, {0x040F0702},	/* 309 */
+  {0x040F0702}, {0x04070F02}, {0x04070F02}, {0x040D0A02}, {0x040D0A02},	/* 314 */
+  {0x040A0D02}, {0x040A0D02}, {0x040C0B02}, {0x040C0B02}, {0x040F0602},	/* 319 */
+  {0x040F0602}, {0x050E0A02}, {0x050F0001}, {0xFF000004}, {0x030B0C02},	/* 324 */
+  {0x030B0C02}, {0x03060F02}, {0x03060F02}, {0x040E0802}, {0x04080E02},	/* 329 */
+  {0x040F0502}, {0x040D0902}, {0x03050F02}, {0x03050F02}, {0x030E0702},	/* 334 */
+  {0x030E0702}, {0x03070E02}, {0x03070E02}, {0x030C0A02}, {0x030C0A02},	/* 339 */
+  {0xFF000004}, {0x030A0C02}, {0x030A0C02}, {0x030B0B02}, {0x030B0B02},	/* 344 */
+  {0x04090D02}, {0x040D0802}, {0x030F0402}, {0x030F0402}, {0x03040F02},	/* 349 */
+  {0x03040F02}, {0x030F0302}, {0x030F0302}, {0x03030F02}, {0x03030F02},	/* 354 */
+  {0x03080D02}, {0x03080D02}, {0xFF000004}, {0x03060E02}, {0x03060E02},	/* 359 */
+  {0x030F0202}, {0x030F0202}, {0x03020F02}, {0x03020F02}, {0x040E0602},	/* 364 */
+  {0x04000F01}, {0x030F0102}, {0x030F0102}, {0x03010F02}, {0x03010F02},	/* 369 */
+  {0x030C0902}, {0x030C0902}, {0x03090C02}, {0x03090C02}, {0xFF000003},	/* 374 */
+  {0x030E0502}, {0x030B0A02}, {0x030A0B02}, {0x03050E02}, {0x030D0702},	/* 379 */
+  {0x03070D02}, {0x030E0402}, {0x03040E02}, {0xFF000003}, {0x030C0802},	/* 384 */
+  {0x03080C02}, {0x030E0302}, {0x030D0602}, {0x03060D02}, {0x03030E02},	/* 389 */
+  {0x030B0902}, {0x03090B02}, {0xFF000004}, {0x030E0202}, {0x030E0202},	/* 394 */
+  {0x030A0A02}, {0x030A0A02}, {0x03020E02}, {0x03020E02}, {0x030E0102},	/* 399 */
+  {0x030E0102}, {0x03010E02}, {0x03010E02}, {0x040E0001}, {0x04000E01},	/* 404 */
+  {0x030D0502}, {0x030D0502}, {0x03050D02}, {0x03050D02}, {0xFF000003},	/* 409 */
+  {0x030C0702}, {0x03070C02}, {0x030D0402}, {0x030B0802}, {0x02040D02},	/* 414 */
+  {0x02040D02}, {0x03080B02}, {0x030A0902}, {0xFF000003}, {0x03090A02},	/* 419 */
+  {0x030C0602}, {0x03060C02}, {0x030D0302}, {0x02030D02}, {0x02030D02},	/* 424 */
+  {0x02020D02}, {0x02020D02}, {0xFF000003}, {0x030D0202}, {0x030D0001},	/* 429 */
+  {0x020D0102}, {0x020D0102}, {0x020B0702}, {0x020B0702}, {0x02070B02},	/* 434 */
+  {0x02070B02}, {0xFF000003}, {0x02010D02}, {0x02010D02}, {0x030C0502},	/* 439 */
+  {0x03000D01}, {0x02050C02}, {0x02050C02}, {0x020A0802}, {0x020A0802},	/* 444 */
+  {0xFF000002}, {0x02080A02}, {0x020C0402}, {0x02040C02}, {0x020B0602},	/* 449 */
+  {0xFF000003}, {0x02060B02}, {0x02060B02}, {0x03090902}, {0x030C0001},	/* 454 */
+  {0x020C0302}, {0x020C0302}, {0x02030C02}, {0x02030C02}, {0xFF000003},	/* 459 */
+  {0x020A0702}, {0x020A0702}, {0x02070A02}, {0x02070A02}, {0x02060A02},	/* 464 */
+  {0x02060A02}, {0x03000C01}, {0x030B0001}, {0xFF000002}, {0x01020C02},	/* 469 */
+  {0x01020C02}, {0x020C0202}, {0x020B0502}, {0xFF000002}, {0x02050B02},	/* 474 */
+  {0x020C0102}, {0x02090802}, {0x02080902}, {0xFF000002}, {0x02010C02},	/* 479 */
+  {0x020B0402}, {0x02040B02}, {0x020A0602}, {0xFF000002}, {0x020B0302},	/* 484 */
+  {0x02090702}, {0x01030B02}, {0x01030B02}, {0xFF000002}, {0x02070902},	/* 489 */
+  {0x02080802}, {0x020B0202}, {0x020A0502}, {0xFF000002}, {0x01020B02},	/* 494 */
+  {0x01020B02}, {0x02050A02}, {0x020B0102}, {0xFF000002}, {0x01010B02},	/* 499 */
+  {0x01010B02}, {0x02000B01}, {0x02090602}, {0xFF000002}, {0x02060902},	/* 504 */
+  {0x020A0402}, {0x02040A02}, {0x02080702}, {0xFF000002}, {0x02070802},	/* 509 */
+  {0x020A0302}, {0x01030A02}, {0x01030A02}, {0xFF000001}, {0x01090502},	/* 514 */
+  {0x01050902}, {0xFF000001}, {0x010A0202}, {0x01020A02}, {0xFF000001},	/* 519 */
+  {0x010A0102}, {0x01010A02}, {0xFF000002}, {0x020A0001}, {0x02000A01},	/* 524 */
+  {0x01080602}, {0x01080602}, {0xFF000001}, {0x01060802}, {0x01090402},	/* 529 */
+  {0xFF000001}, {0x01040902}, {0x01090302}, {0xFF000002}, {0x01030902},	/* 534 */
+  {0x01030902}, {0x02070702}, {0x02090001}, {0xFF000001}, {0x01080502},	/* 539 */
+  {0x01050802}, {0xFF000001}, {0x01090202}, {0x01070602}, {0xFF000001},	/* 544 */
+  {0x01060702}, {0x01020902}, {0xFF000001}, {0x01090102}, {0x01000901},	/* 549 */
+  {0xFF000001}, {0x01080402}, {0x01040802}, {0xFF000001}, {0x01070502},	/* 554 */
+  {0x01050702}, {0xFF000001}, {0x01080302}, {0x01030802}, {0xFF000001},	/* 559 */
+  {0x01060602}, {0x01070402}, {0xFF000001}, {0x01040702}, {0x01080001},	/* 564 */
+  {0xFF000001}, {0x01000801}, {0x01060502}, {0xFF000001}, {0x01050602},	/* 569 */
+  {0x01070302}, {0xFF000001}, {0x01030702}, {0x01060402}, {0xFF000001},	/* 574 */
+  {0x01070001}, {0x01000701}, {0xFF000001}, {0x01060001}, {0x01000601}, /* 579 */
+};
+
+/* max table bits  8 */
+
+/* TABLE 16  256 entries  maxbits 17  linbits  0 */
+static HUFF_ELEMENT huff_table_16[] =
+{
+  {0xFF000008}, {0x00000101}, {0x0000010A}, {0x00000113}, {0x080F0F02},	/*  4 */
+  {0x00000118}, {0x0000011D}, {0x00000120}, {0x08020F02}, {0x00000131},	/*  9 */
+  {0x080F0102}, {0x08010F02}, {0x00000134}, {0x00000145}, {0x00000156},	/* 14 */
+  {0x00000167}, {0x00000178}, {0x00000189}, {0x0000019A}, {0x000001A3},	/* 19 */
+  {0x000001AC}, {0x000001B5}, {0x000001BE}, {0x000001C7}, {0x000001D0},	/* 24 */
+  {0x000001D9}, {0x000001DE}, {0x000001E3}, {0x000001E6}, {0x000001EB},	/* 29 */
+  {0x000001F0}, {0x08010502}, {0x000001F3}, {0x000001F6}, {0x000001F9},	/* 34 */
+  {0x000001FC}, {0x08040102}, {0x08010402}, {0x000001FF}, {0x08030202},	/* 39 */
+  {0x08020302}, {0x07030102}, {0x07030102}, {0x07010302}, {0x07010302},	/* 44 */
+  {0x08030001}, {0x08000301}, {0x07020202}, {0x07020202}, {0x06020102},	/* 49 */
+  {0x06020102}, {0x06020102}, {0x06020102}, {0x06010202}, {0x06010202},	/* 54 */
+  {0x06010202}, {0x06010202}, {0x06020001}, {0x06020001}, {0x06020001},	/* 59 */
+  {0x06020001}, {0x06000201}, {0x06000201}, {0x06000201}, {0x06000201},	/* 64 */
+  {0x04010102}, {0x04010102}, {0x04010102}, {0x04010102}, {0x04010102},	/* 69 */
+  {0x04010102}, {0x04010102}, {0x04010102}, {0x04010102}, {0x04010102},	/* 74 */
+  {0x04010102}, {0x04010102}, {0x04010102}, {0x04010102}, {0x04010102},	/* 79 */
+  {0x04010102}, {0x04010001}, {0x04010001}, {0x04010001}, {0x04010001},	/* 84 */
+  {0x04010001}, {0x04010001}, {0x04010001}, {0x04010001}, {0x04010001},	/* 89 */
+  {0x04010001}, {0x04010001}, {0x04010001}, {0x04010001}, {0x04010001},	/* 94 */
+  {0x04010001}, {0x04010001}, {0x03000101}, {0x03000101}, {0x03000101},	/* 99 */
+  {0x03000101}, {0x03000101}, {0x03000101}, {0x03000101}, {0x03000101},	/* 104 */
+  {0x03000101}, {0x03000101}, {0x03000101}, {0x03000101}, {0x03000101},	/* 109 */
+  {0x03000101}, {0x03000101}, {0x03000101}, {0x03000101}, {0x03000101},	/* 114 */
+  {0x03000101}, {0x03000101}, {0x03000101}, {0x03000101}, {0x03000101},	/* 119 */
+  {0x03000101}, {0x03000101}, {0x03000101}, {0x03000101}, {0x03000101},	/* 124 */
+  {0x03000101}, {0x03000101}, {0x03000101}, {0x03000101}, {0x01000000},	/* 129 */
+  {0x01000000}, {0x01000000}, {0x01000000}, {0x01000000}, {0x01000000},	/* 134 */
+  {0x01000000}, {0x01000000}, {0x01000000}, {0x01000000}, {0x01000000},	/* 139 */
+  {0x01000000}, {0x01000000}, {0x01000000}, {0x01000000}, {0x01000000},	/* 144 */
+  {0x01000000}, {0x01000000}, {0x01000000}, {0x01000000}, {0x01000000},	/* 149 */
+  {0x01000000}, {0x01000000}, {0x01000000}, {0x01000000}, {0x01000000},	/* 154 */
+  {0x01000000}, {0x01000000}, {0x01000000}, {0x01000000}, {0x01000000},	/* 159 */
+  {0x01000000}, {0x01000000}, {0x01000000}, {0x01000000}, {0x01000000},	/* 164 */
+  {0x01000000}, {0x01000000}, {0x01000000}, {0x01000000}, {0x01000000},	/* 169 */
+  {0x01000000}, {0x01000000}, {0x01000000}, {0x01000000}, {0x01000000},	/* 174 */
+  {0x01000000}, {0x01000000}, {0x01000000}, {0x01000000}, {0x01000000},	/* 179 */
+  {0x01000000}, {0x01000000}, {0x01000000}, {0x01000000}, {0x01000000},	/* 184 */
+  {0x01000000}, {0x01000000}, {0x01000000}, {0x01000000}, {0x01000000},	/* 189 */
+  {0x01000000}, {0x01000000}, {0x01000000}, {0x01000000}, {0x01000000},	/* 194 */
+  {0x01000000}, {0x01000000}, {0x01000000}, {0x01000000}, {0x01000000},	/* 199 */
+  {0x01000000}, {0x01000000}, {0x01000000}, {0x01000000}, {0x01000000},	/* 204 */
+  {0x01000000}, {0x01000000}, {0x01000000}, {0x01000000}, {0x01000000},	/* 209 */
+  {0x01000000}, {0x01000000}, {0x01000000}, {0x01000000}, {0x01000000},	/* 214 */
+  {0x01000000}, {0x01000000}, {0x01000000}, {0x01000000}, {0x01000000},	/* 219 */
+  {0x01000000}, {0x01000000}, {0x01000000}, {0x01000000}, {0x01000000},	/* 224 */
+  {0x01000000}, {0x01000000}, {0x01000000}, {0x01000000}, {0x01000000},	/* 229 */
+  {0x01000000}, {0x01000000}, {0x01000000}, {0x01000000}, {0x01000000},	/* 234 */
+  {0x01000000}, {0x01000000}, {0x01000000}, {0x01000000}, {0x01000000},	/* 239 */
+  {0x01000000}, {0x01000000}, {0x01000000}, {0x01000000}, {0x01000000},	/* 244 */
+  {0x01000000}, {0x01000000}, {0x01000000}, {0x01000000}, {0x01000000},	/* 249 */
+  {0x01000000}, {0x01000000}, {0x01000000}, {0x01000000}, {0x01000000},	/* 254 */
+  {0x01000000}, {0x01000000}, {0xFF000003}, {0x030F0E02}, {0x030E0F02},	/* 259 */
+  {0x030F0D02}, {0x030D0F02}, {0x030F0C02}, {0x030C0F02}, {0x030F0B02},	/* 264 */
+  {0x030B0F02}, {0xFF000003}, {0x020F0A02}, {0x020F0A02}, {0x030A0F02},	/* 269 */
+  {0x030F0902}, {0x03090F02}, {0x03080F02}, {0x020F0802}, {0x020F0802},	/* 274 */
+  {0xFF000002}, {0x020F0702}, {0x02070F02}, {0x020F0602}, {0x02060F02},	/* 279 */
+  {0xFF000002}, {0x020F0502}, {0x02050F02}, {0x010F0402}, {0x010F0402},	/* 284 */
+  {0xFF000001}, {0x01040F02}, {0x01030F02}, {0xFF000004}, {0x01000F01},	/* 289 */
+  {0x01000F01}, {0x01000F01}, {0x01000F01}, {0x01000F01}, {0x01000F01},	/* 294 */
+  {0x01000F01}, {0x01000F01}, {0x020F0302}, {0x020F0302}, {0x020F0302},	/* 299 */
+  {0x020F0302}, {0x000000E2}, {0x000000F3}, {0x000000FC}, {0x00000105},	/* 304 */
+  {0xFF000001}, {0x010F0202}, {0x010F0001}, {0xFF000004}, {0x000000FA},	/* 309 */
+  {0x000000FF}, {0x00000104}, {0x00000109}, {0x0000010C}, {0x00000111},	/* 314 */
+  {0x00000116}, {0x00000119}, {0x0000011E}, {0x00000123}, {0x00000128},	/* 319 */
+  {0x04030E02}, {0x0000012D}, {0x00000130}, {0x00000133}, {0x00000136},	/* 324 */
+  {0xFF000004}, {0x00000128}, {0x0000012B}, {0x0000012E}, {0x040D0001},	/* 329 */
+  {0x00000131}, {0x00000134}, {0x00000137}, {0x040C0302}, {0x0000013A},	/* 334 */
+  {0x040C0102}, {0x04000C01}, {0x0000013D}, {0x03020E02}, {0x03020E02},	/* 339 */
+  {0x040E0202}, {0x040E0102}, {0xFF000004}, {0x04030D02}, {0x040D0202},	/* 344 */
+  {0x04020D02}, {0x04010D02}, {0x040B0302}, {0x0000012F}, {0x030D0102},	/* 349 */
+  {0x030D0102}, {0x04040C02}, {0x040B0602}, {0x04030C02}, {0x04070A02},	/* 354 */
+  {0x030C0202}, {0x030C0202}, {0x04020C02}, {0x04050B02}, {0xFF000004},	/* 359 */
+  {0x04010C02}, {0x040C0001}, {0x040B0402}, {0x04040B02}, {0x040A0602},	/* 364 */
+  {0x04060A02}, {0x03030B02}, {0x03030B02}, {0x040A0502}, {0x04050A02},	/* 369 */
+  {0x030B0202}, {0x030B0202}, {0x03020B02}, {0x03020B02}, {0x030B0102},	/* 374 */
+  {0x030B0102}, {0xFF000004}, {0x03010B02}, {0x03010B02}, {0x040B0001},	/* 379 */
+  {0x04000B01}, {0x04090602}, {0x04060902}, {0x040A0402}, {0x04040A02},	/* 384 */
+  {0x04080702}, {0x04070802}, {0x03030A02}, {0x03030A02}, {0x040A0302},	/* 389 */
+  {0x04090502}, {0x030A0202}, {0x030A0202}, {0xFF000004}, {0x04050902},	/* 394 */
+  {0x04080602}, {0x03010A02}, {0x03010A02}, {0x04060802}, {0x04070702},	/* 399 */
+  {0x03040902}, {0x03040902}, {0x04090402}, {0x04070502}, {0x03070602},	/* 404 */
+  {0x03070602}, {0x02020A02}, {0x02020A02}, {0x02020A02}, {0x02020A02},	/* 409 */
+  {0xFF000003}, {0x020A0102}, {0x020A0102}, {0x030A0001}, {0x03000A01},	/* 414 */
+  {0x03090302}, {0x03030902}, {0x03080502}, {0x03050802}, {0xFF000003},	/* 419 */
+  {0x02090202}, {0x02090202}, {0x02020902}, {0x02020902}, {0x03060702},	/* 424 */
+  {0x03090001}, {0x02090102}, {0x02090102}, {0xFF000003}, {0x02010902},	/* 429 */
+  {0x02010902}, {0x03000901}, {0x03080402}, {0x03040802}, {0x03050702},	/* 434 */
+  {0x03080302}, {0x03030802}, {0xFF000003}, {0x03060602}, {0x03080202},	/* 439 */
+  {0x02020802}, {0x02020802}, {0x03070402}, {0x03040702}, {0x02080102},	/* 444 */
+  {0x02080102}, {0xFF000003}, {0x02010802}, {0x02010802}, {0x02000801},	/* 449 */
+  {0x02000801}, {0x03080001}, {0x03060502}, {0x02070302}, {0x02070302},	/* 454 */
+  {0xFF000003}, {0x02030702}, {0x02030702}, {0x03050602}, {0x03060402},	/* 459 */
+  {0x02070202}, {0x02070202}, {0x02020702}, {0x02020702}, {0xFF000003},	/* 464 */
+  {0x03040602}, {0x03050502}, {0x02070001}, {0x02070001}, {0x01070102},	/* 469 */
+  {0x01070102}, {0x01070102}, {0x01070102}, {0xFF000002}, {0x01010702},	/* 474 */
+  {0x01010702}, {0x02000701}, {0x02060302}, {0xFF000002}, {0x02030602},	/* 479 */
+  {0x02050402}, {0x02040502}, {0x02060202}, {0xFF000001}, {0x01020602},	/* 484 */
+  {0x01060102}, {0xFF000002}, {0x01010602}, {0x01010602}, {0x02060001},	/* 489 */
+  {0x02000601}, {0xFF000002}, {0x01030502}, {0x01030502}, {0x02050302},	/* 494 */
+  {0x02040402}, {0xFF000001}, {0x01050202}, {0x01020502}, {0xFF000001},	/* 499 */
+  {0x01050102}, {0x01050001}, {0xFF000001}, {0x01040302}, {0x01030402},	/* 504 */
+  {0xFF000001}, {0x01000501}, {0x01040202}, {0xFF000001}, {0x01020402},	/* 509 */
+  {0x01030302}, {0xFF000001}, {0x01040001}, {0x01000401}, {0xFF000004},	/* 514 */
+  {0x040E0C02}, {0x00000086}, {0x030E0D02}, {0x030E0D02}, {0x03090E02},	/* 519 */
+  {0x03090E02}, {0x040A0E02}, {0x04090D02}, {0x020E0E02}, {0x020E0E02},	/* 524 */
+  {0x020E0E02}, {0x020E0E02}, {0x030D0E02}, {0x030D0E02}, {0x030B0E02},	/* 529 */
+  {0x030B0E02}, {0xFF000003}, {0x020E0B02}, {0x020E0B02}, {0x020D0C02},	/* 534 */
+  {0x020D0C02}, {0x030C0D02}, {0x030B0D02}, {0x020E0A02}, {0x020E0A02},	/* 539 */
+  {0xFF000003}, {0x020C0C02}, {0x020C0C02}, {0x030D0A02}, {0x030A0D02},	/* 544 */
+  {0x030E0702}, {0x030C0A02}, {0x020A0C02}, {0x020A0C02}, {0xFF000003},	/* 549 */
+  {0x03090C02}, {0x030D0702}, {0x020E0502}, {0x020E0502}, {0x010D0B02},	/* 554 */
+  {0x010D0B02}, {0x010D0B02}, {0x010D0B02}, {0xFF000002}, {0x010E0902},	/* 559 */
+  {0x010E0902}, {0x020C0B02}, {0x020B0C02}, {0xFF000002}, {0x020E0802},	/* 564 */
+  {0x02080E02}, {0x020D0902}, {0x02070E02}, {0xFF000002}, {0x020B0B02},	/* 569 */
+  {0x020D0802}, {0x02080D02}, {0x020E0602}, {0xFF000001}, {0x01060E02},	/* 574 */
+  {0x010C0902}, {0xFF000002}, {0x020B0A02}, {0x020A0B02}, {0x02050E02},	/* 579 */
+  {0x02070D02}, {0xFF000002}, {0x010E0402}, {0x010E0402}, {0x02040E02},	/* 584 */
+  {0x020C0802}, {0xFF000001}, {0x01080C02}, {0x010E0302}, {0xFF000002},	/* 589 */
+  {0x010D0602}, {0x010D0602}, {0x02060D02}, {0x020B0902}, {0xFF000002},	/* 594 */
+  {0x02090B02}, {0x020A0A02}, {0x01010E02}, {0x01010E02}, {0xFF000002},	/* 599 */
+  {0x01040D02}, {0x01040D02}, {0x02080B02}, {0x02090A02}, {0xFF000002},	/* 604 */
+  {0x010B0702}, {0x010B0702}, {0x02070B02}, {0x02000D01}, {0xFF000001},	/* 609 */
+  {0x010E0001}, {0x01000E01}, {0xFF000001}, {0x010D0502}, {0x01050D02},	/* 614 */
+  {0xFF000001}, {0x010C0702}, {0x01070C02}, {0xFF000001}, {0x010D0402},	/* 619 */
+  {0x010B0802}, {0xFF000001}, {0x010A0902}, {0x010C0602}, {0xFF000001},	/* 624 */
+  {0x01060C02}, {0x010D0302}, {0xFF000001}, {0x010C0502}, {0x01050C02},	/* 629 */
+  {0xFF000001}, {0x010A0802}, {0x01080A02}, {0xFF000001}, {0x01090902},	/* 634 */
+  {0x010C0402}, {0xFF000001}, {0x01060B02}, {0x010A0702}, {0xFF000001},	/* 639 */
+  {0x010B0502}, {0x01090802}, {0xFF000001}, {0x01080902}, {0x01090702},	/* 644 */
+  {0xFF000001}, {0x01070902}, {0x01080802}, {0xFF000001}, {0x010C0E02},	/* 649 */
+  {0x010D0D02},
+};
+
+/* max table bits  8 */
+/* NO XING TABLE 17 */
+/* NO XING TABLE 18 */
+/* NO XING TABLE 19 */
+/* NO XING TABLE 20 */
+/* NO XING TABLE 21 */
+/* NO XING TABLE 22 */
+/* NO XING TABLE 23 */
+
+/* TABLE 24  256 entries  maxbits 12  linbits  0 */
+static HUFF_ELEMENT huff_table_24[] =
+{
+  {0xFF000009}, {0x080F0E02}, {0x080F0E02}, {0x080E0F02}, {0x080E0F02},	/*  4 */
+  {0x080F0D02}, {0x080F0D02}, {0x080D0F02}, {0x080D0F02}, {0x080F0C02},	/*  9 */
+  {0x080F0C02}, {0x080C0F02}, {0x080C0F02}, {0x080F0B02}, {0x080F0B02},	/* 14 */
+  {0x080B0F02}, {0x080B0F02}, {0x070A0F02}, {0x070A0F02}, {0x070A0F02},	/* 19 */
+  {0x070A0F02}, {0x080F0A02}, {0x080F0A02}, {0x080F0902}, {0x080F0902},	/* 24 */
+  {0x07090F02}, {0x07090F02}, {0x07090F02}, {0x07090F02}, {0x07080F02},	/* 29 */
+  {0x07080F02}, {0x07080F02}, {0x07080F02}, {0x080F0802}, {0x080F0802},	/* 34 */
+  {0x080F0702}, {0x080F0702}, {0x07070F02}, {0x07070F02}, {0x07070F02},	/* 39 */
+  {0x07070F02}, {0x070F0602}, {0x070F0602}, {0x070F0602}, {0x070F0602},	/* 44 */
+  {0x07060F02}, {0x07060F02}, {0x07060F02}, {0x07060F02}, {0x070F0502},	/* 49 */
+  {0x070F0502}, {0x070F0502}, {0x070F0502}, {0x07050F02}, {0x07050F02},	/* 54 */
+  {0x07050F02}, {0x07050F02}, {0x070F0402}, {0x070F0402}, {0x070F0402},	/* 59 */
+  {0x070F0402}, {0x07040F02}, {0x07040F02}, {0x07040F02}, {0x07040F02},	/* 64 */
+  {0x070F0302}, {0x070F0302}, {0x070F0302}, {0x070F0302}, {0x07030F02},	/* 69 */
+  {0x07030F02}, {0x07030F02}, {0x07030F02}, {0x070F0202}, {0x070F0202},	/* 74 */
+  {0x070F0202}, {0x070F0202}, {0x07020F02}, {0x07020F02}, {0x07020F02},	/* 79 */
+  {0x07020F02}, {0x07010F02}, {0x07010F02}, {0x07010F02}, {0x07010F02},	/* 84 */
+  {0x080F0102}, {0x080F0102}, {0x08000F01}, {0x08000F01}, {0x090F0001},	/* 89 */
+  {0x00000201}, {0x00000206}, {0x0000020B}, {0x00000210}, {0x00000215},	/* 94 */
+  {0x0000021A}, {0x0000021F}, {0x040F0F02}, {0x040F0F02}, {0x040F0F02},	/* 99 */
+  {0x040F0F02}, {0x040F0F02}, {0x040F0F02}, {0x040F0F02}, {0x040F0F02},	/* 104 */
+  {0x040F0F02}, {0x040F0F02}, {0x040F0F02}, {0x040F0F02}, {0x040F0F02},	/* 109 */
+  {0x040F0F02}, {0x040F0F02}, {0x040F0F02}, {0x040F0F02}, {0x040F0F02},	/* 114 */
+  {0x040F0F02}, {0x040F0F02}, {0x040F0F02}, {0x040F0F02}, {0x040F0F02},	/* 119 */
+  {0x040F0F02}, {0x040F0F02}, {0x040F0F02}, {0x040F0F02}, {0x040F0F02},	/* 124 */
+  {0x040F0F02}, {0x040F0F02}, {0x040F0F02}, {0x040F0F02}, {0x00000224},	/* 129 */
+  {0x00000229}, {0x00000232}, {0x00000237}, {0x0000023A}, {0x0000023F},	/* 134 */
+  {0x00000242}, {0x00000245}, {0x0000024A}, {0x0000024D}, {0x00000250},	/* 139 */
+  {0x00000253}, {0x00000256}, {0x00000259}, {0x0000025C}, {0x0000025F},	/* 144 */
+  {0x00000262}, {0x00000265}, {0x00000268}, {0x0000026B}, {0x0000026E},	/* 149 */
+  {0x00000271}, {0x00000274}, {0x00000277}, {0x0000027A}, {0x0000027D},	/* 154 */
+  {0x00000280}, {0x00000283}, {0x00000288}, {0x0000028B}, {0x0000028E},	/* 159 */
+  {0x00000291}, {0x00000294}, {0x00000297}, {0x0000029A}, {0x0000029F},	/* 164 */
+  {0x09040B02}, {0x000002A4}, {0x000002A7}, {0x000002AA}, {0x09030B02},	/* 169 */
+  {0x09080802}, {0x000002AF}, {0x09020B02}, {0x000002B2}, {0x000002B5},	/* 174 */
+  {0x09060902}, {0x09040A02}, {0x000002B8}, {0x09070802}, {0x090A0302},	/* 179 */
+  {0x09030A02}, {0x09090502}, {0x09050902}, {0x090A0202}, {0x09020A02},	/* 184 */
+  {0x09010A02}, {0x09080602}, {0x09060802}, {0x09070702}, {0x09090402},	/* 189 */
+  {0x09040902}, {0x09090302}, {0x09030902}, {0x09080502}, {0x09050802},	/* 194 */
+  {0x09090202}, {0x09070602}, {0x09060702}, {0x09020902}, {0x09090102},	/* 199 */
+  {0x09010902}, {0x09080402}, {0x09040802}, {0x09070502}, {0x09050702},	/* 204 */
+  {0x09080302}, {0x09030802}, {0x09060602}, {0x09080202}, {0x09020802},	/* 209 */
+  {0x09080102}, {0x09070402}, {0x09040702}, {0x09010802}, {0x000002BB},	/* 214 */
+  {0x09060502}, {0x09050602}, {0x09070102}, {0x000002BE}, {0x08030702},	/* 219 */
+  {0x08030702}, {0x09070302}, {0x09070202}, {0x08020702}, {0x08020702},	/* 224 */
+  {0x08060402}, {0x08060402}, {0x08040602}, {0x08040602}, {0x08050502},	/* 229 */
+  {0x08050502}, {0x08010702}, {0x08010702}, {0x08060302}, {0x08060302},	/* 234 */
+  {0x08030602}, {0x08030602}, {0x08050402}, {0x08050402}, {0x08040502},	/* 239 */
+  {0x08040502}, {0x08060202}, {0x08060202}, {0x08020602}, {0x08020602},	/* 244 */
+  {0x08060102}, {0x08060102}, {0x08010602}, {0x08010602}, {0x09060001},	/* 249 */
+  {0x09000601}, {0x08050302}, {0x08050302}, {0x08030502}, {0x08030502},	/* 254 */
+  {0x08040402}, {0x08040402}, {0x08050202}, {0x08050202}, {0x08020502},	/* 259 */
+  {0x08020502}, {0x08050102}, {0x08050102}, {0x09050001}, {0x09000501},	/* 264 */
+  {0x07010502}, {0x07010502}, {0x07010502}, {0x07010502}, {0x08040302},	/* 269 */
+  {0x08040302}, {0x08030402}, {0x08030402}, {0x07040202}, {0x07040202},	/* 274 */
+  {0x07040202}, {0x07040202}, {0x07020402}, {0x07020402}, {0x07020402},	/* 279 */
+  {0x07020402}, {0x07030302}, {0x07030302}, {0x07030302}, {0x07030302},	/* 284 */
+  {0x07040102}, {0x07040102}, {0x07040102}, {0x07040102}, {0x07010402},	/* 289 */
+  {0x07010402}, {0x07010402}, {0x07010402}, {0x08040001}, {0x08040001},	/* 294 */
+  {0x08000401}, {0x08000401}, {0x07030202}, {0x07030202}, {0x07030202},	/* 299 */
+  {0x07030202}, {0x07020302}, {0x07020302}, {0x07020302}, {0x07020302},	/* 304 */
+  {0x06030102}, {0x06030102}, {0x06030102}, {0x06030102}, {0x06030102},	/* 309 */
+  {0x06030102}, {0x06030102}, {0x06030102}, {0x06010302}, {0x06010302},	/* 314 */
+  {0x06010302}, {0x06010302}, {0x06010302}, {0x06010302}, {0x06010302},	/* 319 */
+  {0x06010302}, {0x07030001}, {0x07030001}, {0x07030001}, {0x07030001},	/* 324 */
+  {0x07000301}, {0x07000301}, {0x07000301}, {0x07000301}, {0x06020202},	/* 329 */
+  {0x06020202}, {0x06020202}, {0x06020202}, {0x06020202}, {0x06020202},	/* 334 */
+  {0x06020202}, {0x06020202}, {0x05020102}, {0x05020102}, {0x05020102},	/* 339 */
+  {0x05020102}, {0x05020102}, {0x05020102}, {0x05020102}, {0x05020102},	/* 344 */
+  {0x05020102}, {0x05020102}, {0x05020102}, {0x05020102}, {0x05020102},	/* 349 */
+  {0x05020102}, {0x05020102}, {0x05020102}, {0x05010202}, {0x05010202},	/* 354 */
+  {0x05010202}, {0x05010202}, {0x05010202}, {0x05010202}, {0x05010202},	/* 359 */
+  {0x05010202}, {0x05010202}, {0x05010202}, {0x05010202}, {0x05010202},	/* 364 */
+  {0x05010202}, {0x05010202}, {0x05010202}, {0x05010202}, {0x06020001},	/* 369 */
+  {0x06020001}, {0x06020001}, {0x06020001}, {0x06020001}, {0x06020001},	/* 374 */
+  {0x06020001}, {0x06020001}, {0x06000201}, {0x06000201}, {0x06000201},	/* 379 */
+  {0x06000201}, {0x06000201}, {0x06000201}, {0x06000201}, {0x06000201},	/* 384 */
+  {0x04010102}, {0x04010102}, {0x04010102}, {0x04010102}, {0x04010102},	/* 389 */
+  {0x04010102}, {0x04010102}, {0x04010102}, {0x04010102}, {0x04010102},	/* 394 */
+  {0x04010102}, {0x04010102}, {0x04010102}, {0x04010102}, {0x04010102},	/* 399 */
+  {0x04010102}, {0x04010102}, {0x04010102}, {0x04010102}, {0x04010102},	/* 404 */
+  {0x04010102}, {0x04010102}, {0x04010102}, {0x04010102}, {0x04010102},	/* 409 */
+  {0x04010102}, {0x04010102}, {0x04010102}, {0x04010102}, {0x04010102},	/* 414 */
+  {0x04010102}, {0x04010102}, {0x04010001}, {0x04010001}, {0x04010001},	/* 419 */
+  {0x04010001}, {0x04010001}, {0x04010001}, {0x04010001}, {0x04010001},	/* 424 */
+  {0x04010001}, {0x04010001}, {0x04010001}, {0x04010001}, {0x04010001},	/* 429 */
+  {0x04010001}, {0x04010001}, {0x04010001}, {0x04010001}, {0x04010001},	/* 434 */
+  {0x04010001}, {0x04010001}, {0x04010001}, {0x04010001}, {0x04010001},	/* 439 */
+  {0x04010001}, {0x04010001}, {0x04010001}, {0x04010001}, {0x04010001},	/* 444 */
+  {0x04010001}, {0x04010001}, {0x04010001}, {0x04010001}, {0x04000101},	/* 449 */
+  {0x04000101}, {0x04000101}, {0x04000101}, {0x04000101}, {0x04000101},	/* 454 */
+  {0x04000101}, {0x04000101}, {0x04000101}, {0x04000101}, {0x04000101},	/* 459 */
+  {0x04000101}, {0x04000101}, {0x04000101}, {0x04000101}, {0x04000101},	/* 464 */
+  {0x04000101}, {0x04000101}, {0x04000101}, {0x04000101}, {0x04000101},	/* 469 */
+  {0x04000101}, {0x04000101}, {0x04000101}, {0x04000101}, {0x04000101},	/* 474 */
+  {0x04000101}, {0x04000101}, {0x04000101}, {0x04000101}, {0x04000101},	/* 479 */
+  {0x04000101}, {0x04000000}, {0x04000000}, {0x04000000}, {0x04000000},	/* 484 */
+  {0x04000000}, {0x04000000}, {0x04000000}, {0x04000000}, {0x04000000},	/* 489 */
+  {0x04000000}, {0x04000000}, {0x04000000}, {0x04000000}, {0x04000000},	/* 494 */
+  {0x04000000}, {0x04000000}, {0x04000000}, {0x04000000}, {0x04000000},	/* 499 */
+  {0x04000000}, {0x04000000}, {0x04000000}, {0x04000000}, {0x04000000},	/* 504 */
+  {0x04000000}, {0x04000000}, {0x04000000}, {0x04000000}, {0x04000000},	/* 509 */
+  {0x04000000}, {0x04000000}, {0x04000000}, {0xFF000002}, {0x020E0E02},	/* 514 */
+  {0x020E0D02}, {0x020D0E02}, {0x020E0C02}, {0xFF000002}, {0x020C0E02},	/* 519 */
+  {0x020D0D02}, {0x020E0B02}, {0x020B0E02}, {0xFF000002}, {0x020D0C02},	/* 524 */
+  {0x020C0D02}, {0x020E0A02}, {0x020A0E02}, {0xFF000002}, {0x020D0B02},	/* 529 */
+  {0x020B0D02}, {0x020C0C02}, {0x020E0902}, {0xFF000002}, {0x02090E02},	/* 534 */
+  {0x020D0A02}, {0x020A0D02}, {0x020C0B02}, {0xFF000002}, {0x020B0C02},	/* 539 */
+  {0x020E0802}, {0x02080E02}, {0x020D0902}, {0xFF000002}, {0x02090D02},	/* 544 */
+  {0x020E0702}, {0x02070E02}, {0x020C0A02}, {0xFF000002}, {0x020A0C02},	/* 549 */
+  {0x020B0B02}, {0x020D0802}, {0x02080D02}, {0xFF000003}, {0x030E0001},	/* 554 */
+  {0x03000E01}, {0x020D0001}, {0x020D0001}, {0x01060E02}, {0x01060E02},	/* 559 */
+  {0x01060E02}, {0x01060E02}, {0xFF000002}, {0x020E0602}, {0x020C0902},	/* 564 */
+  {0x01090C02}, {0x01090C02}, {0xFF000001}, {0x010E0502}, {0x010A0B02},	/* 569 */
+  {0xFF000002}, {0x01050E02}, {0x01050E02}, {0x020B0A02}, {0x020D0702},	/* 574 */
+  {0xFF000001}, {0x01070D02}, {0x01040E02}, {0xFF000001}, {0x010C0802},	/* 579 */
+  {0x01080C02}, {0xFF000002}, {0x020E0402}, {0x020E0202}, {0x010E0302},	/* 584 */
+  {0x010E0302}, {0xFF000001}, {0x010D0602}, {0x01060D02}, {0xFF000001},	/* 589 */
+  {0x01030E02}, {0x010B0902}, {0xFF000001}, {0x01090B02}, {0x010A0A02},	/* 594 */
+  {0xFF000001}, {0x01020E02}, {0x010E0102}, {0xFF000001}, {0x01010E02},	/* 599 */
+  {0x010D0502}, {0xFF000001}, {0x01050D02}, {0x010C0702}, {0xFF000001},	/* 604 */
+  {0x01070C02}, {0x010D0402}, {0xFF000001}, {0x010B0802}, {0x01080B02},	/* 609 */
+  {0xFF000001}, {0x01040D02}, {0x010A0902}, {0xFF000001}, {0x01090A02},	/* 614 */
+  {0x010C0602}, {0xFF000001}, {0x01060C02}, {0x010D0302}, {0xFF000001},	/* 619 */
+  {0x01030D02}, {0x010D0202}, {0xFF000001}, {0x01020D02}, {0x010D0102},	/* 624 */
+  {0xFF000001}, {0x010B0702}, {0x01070B02}, {0xFF000001}, {0x01010D02},	/* 629 */
+  {0x010C0502}, {0xFF000001}, {0x01050C02}, {0x010A0802}, {0xFF000001},	/* 634 */
+  {0x01080A02}, {0x01090902}, {0xFF000001}, {0x010C0402}, {0x01040C02},	/* 639 */
+  {0xFF000001}, {0x010B0602}, {0x01060B02}, {0xFF000002}, {0x02000D01},	/* 644 */
+  {0x020C0001}, {0x010C0302}, {0x010C0302}, {0xFF000001}, {0x01030C02},	/* 649 */
+  {0x010A0702}, {0xFF000001}, {0x01070A02}, {0x010C0202}, {0xFF000001},	/* 654 */
+  {0x01020C02}, {0x010B0502}, {0xFF000001}, {0x01050B02}, {0x010C0102},	/* 659 */
+  {0xFF000001}, {0x01090802}, {0x01080902}, {0xFF000001}, {0x01010C02},	/* 664 */
+  {0x010B0402}, {0xFF000002}, {0x02000C01}, {0x020B0001}, {0x010B0302},	/* 669 */
+  {0x010B0302}, {0xFF000002}, {0x02000B01}, {0x020A0001}, {0x010A0102},	/* 674 */
+  {0x010A0102}, {0xFF000001}, {0x010A0602}, {0x01060A02}, {0xFF000001},	/* 679 */
+  {0x01090702}, {0x01070902}, {0xFF000002}, {0x02000A01}, {0x02090001},	/* 684 */
+  {0x01000901}, {0x01000901}, {0xFF000001}, {0x010B0202}, {0x010A0502},	/* 689 */
+  {0xFF000001}, {0x01050A02}, {0x010B0102}, {0xFF000001}, {0x01010B02},	/* 694 */
+  {0x01090602}, {0xFF000001}, {0x010A0402}, {0x01080702}, {0xFF000001},	/* 699 */
+  {0x01080001}, {0x01000801}, {0xFF000001}, {0x01070001}, {0x01000701}, /* 704 */
+};
+
+/* max table bits  9 */
+/* NO XING TABLE 25 */
+/* NO XING TABLE 26 */
+/* NO XING TABLE 27 */
+/* NO XING TABLE 28 */
+/* NO XING TABLE 29 */
+/* NO XING TABLE 30 */
+/* NO XING TABLE 31 */
+/* done */

codecs/mp3/include/itype.h

@@ -0,0 +1,108 @@
+/*____________________________________________________________________________
+	
+	FreeAmp - The Free MP3 Player
+
+        MP3 Decoder originally Copyright (C) 1995-1997 Xing Technology
+        Corp.  http://www.xingtech.com
+
+	Portions Copyright (C) 1998 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.
+	
+	$Id$
+____________________________________________________________________________*/
+
+/*---------------------------------------------------------------
+
+mpeg Layer II audio decoder, integer version
+variable type control
+
+-----------------------------------------------------------------*/
+/*-----------------------------------------------------------------
+Variable types can have a large impact on performance.  If the
+native type int is 32 bit or better, setting all variables to the
+native int is probably the best bet.  Machines with fast floating
+point handware will probably run faster with the floating point
+version of this decoder.
+
+On 16 bit machines, use the native 16 bit int where possible
+with special consideration given to the multiplies used in
+the dct and window (see below).
+
+
+The code uses the type INT32 when 32 or more bits are required.
+Use the native int if possible.
+Signed types are required for all but DCTCOEF which may be unsigned.
+
+THe major parts of the decoder are: bit stream unpack (iup.c),
+dct (cidct.c), and window (iwinq.c).  The compute time relationship
+is usually  unpack < dct < window.
+
+-------------------------------------------------------------------*/
+
+/*-------------- dct cidct.c -------------------------------------------
+dct input is type SAMPLEINT, output is WININT
+
+DCTCOEF:  dct coefs, 16 or more bits required
+DCTBITS:  fractional bits in dct coefs. Coefs are unsigned in
+          the range 0.50 to 10.2.  DCTBITS=10 is a compromise
+          between precision and the possibility of overflowing
+          intermediate results.
+
+
+DCTSATURATE:  If set, saturates dct output to 16 bit.
+              Dct output may overflow if WININT is 16 bit, overflow
+              is rare, but very noisy.  Define to 1 for 16 bit WININT.
+              Define to 0 otherwise.
+
+The multiply used in the dct (routine forward_bf in cidct.c) requires
+the multiplication of a 32 bit variable by a 16 bit unsigned coef
+to produce a signed 32 bit result. On 16 bit machines this could be
+faster than a full 32x32 multiply.
+
+------------------------------------------------------------------*/
+/*-------------- WINDOW iwinq.c ---------------------------------------
+window input is type WININT, output is short (16 bit pcm audio)
+
+window coefs WINBITS fractional bits,
+        coefs are signed range in -1.15 to 0.96.
+        WINBITS=14 is maximum for 16 bit signed representation.
+        Some CPU's will multiply faster with fewer bits.
+        WINBITS less that 8 may cause noticeable quality loss.
+
+WINMULT defines the multiply used in the window (iwinq.c)
+WINMULT must produce a 32 bit (or better) result, although both
+multipliers may be 16 bit.  A 16x16-->32 multiply may offer
+a performance advantage if the compiler can be coerced into
+doing the right thing.
+------------------------------------------------------------------*/
+/*-- settings for MS C++ 4.0 flat 32 bit (long=int=32bit) --*/
+/*-- asm replacement modules must use these settings ---*/
+
+typedef long INT32;
+typedef unsigned long UINT32;
+
+typedef int SAMPLEINT;
+
+typedef int DCTCOEF;
+
+#define DCTBITS 10
+#define DCTSATURATE 0
+
+typedef int WININT;
+typedef int WINCOEF;
+
+#define WINBITS 10
+#define WINMULT(x,coef)  ((x)*(coef))

codecs/mp3/include/jdw.h

@@ -0,0 +1,28 @@
+/*____________________________________________________________________________
+	
+	FreeAmp - The Free MP3 Player
+
+	Copyright (C) 1998 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.
+	
+	$Id$
+____________________________________________________________________________*/
+
+/* LOL */
+
+#ifndef min
+#define min(a,b) ((a>b)?b:a)
+#endif

codecs/mp3/include/mhead.h

@@ -0,0 +1,299 @@
+/*____________________________________________________________________________
+	
+	FreeAmp - The Free MP3 Player
+
+        MP3 Decoder originally Copyright (C) 1995-1997 Xing Technology
+        Corp.  http://www.xingtech.com
+
+	Portions Copyright (C) 1998 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.
+	
+	$Id$
+____________________________________________________________________________*/
+
+/* portable copy of eco\mhead.h */
+/* mpeg audio header   */
+
+typedef struct
+{
+   int sync;			/* 1 if valid sync */
+   int id;
+   int option;
+   int prot;
+   int br_index;
+   int sr_index;
+   int pad;
+   int private_bit;
+   int mode;
+   int mode_ext;
+   int cr;
+   int original;
+   int emphasis;
+}
+MPEG_HEAD;
+
+/* portable mpeg audio decoder, decoder functions */
+typedef struct
+{
+   int in_bytes;
+   int out_bytes;
+}
+IN_OUT;
+
+
+typedef struct
+{
+   int channels;
+   int outvalues;
+   long samprate;
+   int bits;
+   int framebytes;
+   int type;
+}
+DEC_INFO;
+
+typedef IN_OUT(*AUDIO_DECODE_ROUTINE) (void *mv, unsigned char *bs, signed short *pcm);
+typedef IN_OUT(*DECODE_FUNCTION) (void *mv, unsigned char *bs, unsigned char *pcm);
+
+struct _mpeg;
+
+typedef struct _mpeg  MPEG;
+
+typedef void (*SBT_FUNCTION_F) (MPEG *m, float *sample, short *pcm, int n);
+
+/* main data bit buffer */
+#define NBUF (8*1024)
+#define BUF_TRIGGER (NBUF-1500)
+
+typedef void (*XFORM_FUNCTION) (void *mv, void *pcm, int igr);
+
+struct _mpeg
+{
+	struct {
+		float look_c_value[18];	/* built by init */
+		unsigned char *bs_ptr;
+		unsigned long bitbuf;
+		int bits;
+		long bitval;
+		int outbytes;
+		int framebytes;
+		int outvalues;
+		int pad;
+		int stereo_sb;
+		DEC_INFO decinfo;		/* global for Layer III */
+		int max_sb;
+		int nsb_limit;
+		int first_pass;
+		int first_pass_L1;
+		int bit_skip;
+		int nbat[4];
+		int bat[4][16];
+		int ballo[64];		/* set by unpack_ba */
+		unsigned int samp_dispatch[66];	/* set by unpack_ba */
+		float c_value[64];	/* set by unpack_ba */
+		unsigned int sf_dispatch[66];	/* set by unpack_ba */
+		float sf_table[64];
+		float cs_factor[3][64];
+		float *sample;		/* global for use by Later 3 */
+		signed char group3_table[32][3];
+		signed char group5_table[128][3];
+		signed short group9_table[1024][3];
+		SBT_FUNCTION_F sbt;
+		AUDIO_DECODE_ROUTINE audio_decode_routine ;
+		float *cs_factorL1;
+		float look_c_valueL1[16];
+		int nbatL1;
+
+	} cup;
+
+	struct {
+		/* cupl3.c */
+		int nBand[2][22];		/* [long/short][cb] */
+		int sfBandIndex[2][22];		/* [long/short][cb] */
+		int mpeg25_flag;
+		int iframe;
+		int band_limit;
+		int band_limit21;
+		int band_limit12;
+		int band_limit_nsb;
+		int nsb_limit;
+		int gaim_adjust;
+		int id;
+		int ncbl_mixed;
+		int gain_adjust;
+		int sr_index;
+		int outvalues;
+		int outbytes;
+		int half_outbytes;
+		int framebytes;
+		int padframebytes;
+		int crcbytes;
+		int pad;
+		int stereo_flag;
+		int nchan;
+		int ms_mode;
+		int is_mode;
+		unsigned int zero_level_pcm;
+		CB_INFO cb_info[2][2];
+		IS_SF_INFO is_sf_info;	/* MPEG-2 intensity stereo */ 
+		unsigned char buf[NBUF];
+		int buf_ptr0;
+		int buf_ptr1;
+		int main_pos_bit;
+		SIDE_INFO side_info;
+		SCALEFACT sf[2][2];	/* [gr][ch] */
+		int nsamp[2][2];		/* must start = 0, for nsamp[igr_prev] */
+		float yout[576];		/* hybrid out, sbt in */
+		SAMPLE sample[2][2][576];
+		SBT_FUNCTION_F sbt_L3;
+		XFORM_FUNCTION Xform;
+		DECODE_FUNCTION decode_function;
+		/* msis.c */
+		/*-- windows by block type --*/
+		float win[4][36];
+		float csa[8][2];		/* antialias */
+		float lr[2][8][2];	/* [ms_mode 0/1][sf][left/right]  */
+		float lr2[2][2][64][2];
+		/* l3dq.c */
+		float look_global[256 + 2 + 4];
+		float look_scale[2][4][32];
+#define ISMAX 32
+		float look_pow[2 * ISMAX];
+		float look_subblock[8];
+		float re_buf[192][3];
+	} cupl;
+	struct {
+		signed int vb_ptr;
+		signed int vb2_ptr;
+		float vbuf[512];
+		float vbuf2[512];
+   		int first_pass;
+	} csbt;
+	struct {
+		float coef32[31];	/* 32 pt dct coefs */
+	} cdct;
+};
+
+typedef int (*CVT_FUNCTION_8) (void *mv, unsigned char *pcm);
+
+typedef struct
+{
+	struct {
+		unsigned char look_u[8192];
+        short pcm[2304];
+        int ncnt;
+        int ncnt1;
+        int nlast;
+        int ndeci;
+        int kdeci;
+		int first_pass;
+        short xsave;
+		CVT_FUNCTION_8 convert_routine;
+	} dec;
+	MPEG cupper;
+}
+MPEG8;
+
+#include "itype.h"
+
+typedef void (*SBT_FUNCTION) (SAMPLEINT * sample, short *pcm, int n);
+typedef void (*UNPACK_FUNCTION) ();
+
+typedef struct
+{
+	struct {
+		DEC_INFO decinfo;
+		int pad;
+		int look_c_value[18];	/* built by init */
+		int look_c_shift[18];	/* built by init */
+		int outbytes;
+		int framebytes;
+		int outvalues;
+		int max_sb;
+		int stereo_sb;
+		int nsb_limit;
+		int bit_skip;
+		int nbat[4];
+		int bat[4][16];
+		int ballo[64];		/* set by unpack_ba */
+		unsigned int samp_dispatch[66];	/* set by unpack_ba */
+		int c_value[64];		/* set by unpack_ba */
+		int c_shift[64];		/* set by unpack_ba */
+		unsigned int sf_dispatch[66];	/* set by unpack_ba */
+		int sf_table[64];
+		INT32 cs_factor[3][64];
+		SAMPLEINT sample[2304];
+		signed char group3_table[32][3];
+		signed char group5_table[128][3];
+		signed short group9_table[1024][3];
+		int nsbt;
+		SBT_FUNCTION sbt;
+		UNPACK_FUNCTION unpack_routine;
+		unsigned char *bs_ptr;
+		UINT32 bitbuf;
+		int bits;
+		INT32 bitval;
+		int first_pass;
+		int first_pass_L1;
+		int nbatL1;
+		INT32 *cs_factorL1;
+		int look_c_valueL1[16]; /* built by init */
+		int look_c_shiftL1[16];	/* built by init */
+	} iup;
+}
+MPEGI;
+
+#ifdef __cplusplus
+extern "C"
+{
+#endif
+
+
+   void mpeg_init(MPEG *m);
+   int head_info(unsigned char *buf, unsigned int n, MPEG_HEAD * h);
+   int head_info2(unsigned char *buf,
+	   unsigned int n, MPEG_HEAD * h, int *br);
+	int head_info3(unsigned char *buf, unsigned int n, MPEG_HEAD *h, int*br, unsigned int *searchForward);
+/* head_info returns framebytes > 0 for success */
+/* audio_decode_init returns 1 for success, 0 for fail */
+/* audio_decode returns in_bytes = 0 on sync loss */
+
+   int audio_decode_init(MPEG *m, MPEG_HEAD * h, int framebytes_arg,
+		   int reduction_code, int transform_code, int convert_code,
+			 int freq_limit);
+   void audio_decode_info(MPEG *m, DEC_INFO * info);
+   IN_OUT audio_decode(MPEG *m, unsigned char *bs, short *pcm);
+
+   void mpeg8_init(MPEG8 *m);
+   int audio_decode8_init(MPEG8 *m, MPEG_HEAD * h, int framebytes_arg,
+		   int reduction_code, int transform_code, int convert_code,
+			  int freq_limit);
+   void audio_decode8_info(MPEG8 *m, DEC_INFO * info);
+   IN_OUT audio_decode8(MPEG8 *m, unsigned char *bs, short *pcmbuf);
+
+/*-- integer decode --*/
+   void i_mpeg_init(MPEGI *m);
+   int i_audio_decode_init(MPEGI *m, MPEG_HEAD * h, int framebytes_arg,
+		   int reduction_code, int transform_code, int convert_code,
+			   int freq_limit);
+   void i_audio_decode_info(MPEGI *m, DEC_INFO * info);
+   IN_OUT i_audio_decode(MPEGI *m, unsigned char *bs, short *pcm);
+
+
+
+#ifdef __cplusplus
+}
+#endif

codecs/mp3/include/port.h

@@ -0,0 +1,99 @@
+/*____________________________________________________________________________
+	
+	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.
+	
+	$Id$
+____________________________________________________________________________*/
+
+#ifndef O_BINARY
+#define O_BINARY 0
+#endif
+
+
+/*--- no kb function unless DOS ---*/
+
+#ifndef KB_OK
+#ifdef __MSDOS__
+#define KB_OK
+#endif
+#ifdef _CONSOLE
+#define KB_OK
+#endif
+#endif
+
+#ifdef NEED_KBHIT
+#ifdef KB_OK
+#ifdef _MSC_VER
+#pragma warning(disable: 4032)
+#endif
+#include <conio.h>
+#else
+static int kbhit()
+{
+   return 0;
+}
+static int getch()
+{
+   return 0;
+}
+#endif
+#endif
+
+/*-- no pcm conversion to wave required 
+ if short = 16 bits and little endian ---*/
+
+/* mods 1/9/97 LITTLE_SHORT16 detect */
+
+#ifndef LITTLE_SHORT16
+  #ifdef __MSDOS__
+    #undef LITTLE_SHORT16
+    #define LITTLE_SHORT16
+  #endif
+  #ifdef WIN32
+    #undef LITTLE_SHORT16
+    #define LITTLE_SHORT16
+  #endif
+  #ifdef _M_IX86
+    #undef LITTLE_SHORT16
+    #define LITTLE_SHORT16
+  #endif
+#endif
+
+
+// JDW //
+//#ifdef LITTLE_SHORT16
+//#define cvt_to_wave_init(a)
+//#define cvt_to_wave(a, b)  b
+//#else
+//void cvt_to_wave_init(int bits);
+//unsigned int cvt_to_wave(void *a, unsigned int b);
+//
+//#endif
+#ifdef LITTLE_SHORT16
+#define cvt_to_wave_init(a)
+#define cvt_to_wave(a, b) b
+#else
+void cvt_to_wave_init(int);
+unsigned int cvt_to_wave(unsigned char *,unsigned int);
+#endif
+
+int cvt_to_wave_test(void);

codecs/mp3/include/protos.h

@@ -0,0 +1,52 @@
+/*====================================================================*/
+int hybrid(MPEG *m, void *xin, void *xprev, float *y,
+	   int btype, int nlong, int ntot, int nprev);
+int hybrid_sum(MPEG *m, void *xin, void *xin_left, float *y,
+	       int btype, int nlong, int ntot);
+void sum_f_bands(void *a, void *b, int n);
+void FreqInvert(float *y, int n);
+void antialias(MPEG *m, void *x, int n);
+void ms_process(void *x, int n);	/* sum-difference stereo */
+void is_process_MPEG1(MPEG *m, void *x,	/* intensity stereo */
+		      SCALEFACT * sf,
+		      CB_INFO cb_info[2],	/* [ch] */
+		      int nsamp, int ms_mode);
+void is_process_MPEG2(MPEG *m, void *x,	/* intensity stereo */
+		      SCALEFACT * sf,
+		      CB_INFO cb_info[2],	/* [ch] */
+		      IS_SF_INFO * is_sf_info,
+		      int nsamp, int ms_mode);
+
+void unpack_huff(void *xy, int n, int ntable);
+int unpack_huff_quad(void *vwxy, int n, int nbits, int ntable);
+void dequant(MPEG *m, SAMPLE sample[], int *nsamp,
+	     SCALEFACT * sf,
+	     GR * gr,
+	     CB_INFO * cb_info, int ncbl_mixed);
+void unpack_sf_sub_MPEG1(SCALEFACT * scalefac, GR * gr,
+			 int scfsi,	/* bit flag */
+			 int igr);
+void unpack_sf_sub_MPEG2(SCALEFACT sf[],	/* return intensity scale */
+			 GR * grdat,
+			 int is_and_ch, IS_SF_INFO * is_sf_info);
+
+
+/*---------- quant ---------------------------------*/
+/* 8 bit lookup x = pow(2.0, 0.25*(global_gain-210)) */
+float *quant_init_global_addr(MPEG *m);
+
+
+/* x = pow(2.0, -0.5*(1+scalefact_scale)*scalefac + preemp) */
+typedef float LS[4][32];
+LS *quant_init_scale_addr(MPEG *m);
+
+
+float *quant_init_pow_addr(MPEG *m);
+float *quant_init_subblock_addr(MPEG *m);
+
+typedef int iARRAY22[22];
+iARRAY22 *quant_init_band_addr(MPEG *m);
+
+/*---------- antialias ---------------------------------*/
+typedef float PAIR[2];
+PAIR *alias_init_addr(MPEG *m);

codecs/mp3/include/tableawd.h

@@ -0,0 +1,93 @@
+/*____________________________________________________________________________
+	
+	FreeAmp - The Free MP3 Player
+
+        MP3 Decoder originally Copyright (C) 1995-1997 Xing Technology
+        Corp.  http://www.xingtech.com
+
+	Portions Copyright (C) 1998 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.
+	
+	$Id$
+____________________________________________________________________________*/
+
+/* decoder analysis window gen by dinit.c (asm version table gen) */
+0.000000000f, 0.000442505f, -0.003250122f, 0.007003784f,
+-0.031082151f, 0.078628540f, -0.100311279f, 0.572036743f,
+-1.144989014f, -0.572036743f, -0.100311279f, -0.078628540f,
+-0.031082151f, -0.007003784f, -0.003250122f, -0.000442505f,
+0.000015259f, 0.000473022f, -0.003326416f, 0.007919312f,
+-0.030517576f, 0.084182739f, -0.090927124f, 0.600219727f,
+-1.144287109f, -0.543823242f, -0.108856201f, -0.073059082f,
+-0.031478882f, -0.006118774f, -0.003173828f, -0.000396729f,
+0.000015259f, 0.000534058f, -0.003387451f, 0.008865356f,
+-0.029785154f, 0.089706421f, -0.080688477f, 0.628295898f,
+-1.142211914f, -0.515609741f, -0.116577141f, -0.067520142f,
+-0.031738281f, -0.005294800f, -0.003082275f, -0.000366211f,
+0.000015259f, 0.000579834f, -0.003433228f, 0.009841919f,
+-0.028884888f, 0.095169067f, -0.069595337f, 0.656219482f,
+-1.138763428f, -0.487472534f, -0.123474121f, -0.061996460f,
+-0.031845093f, -0.004486084f, -0.002990723f, -0.000320435f,
+0.000015259f, 0.000625610f, -0.003463745f, 0.010848999f,
+-0.027801514f, 0.100540161f, -0.057617184f, 0.683914185f,
+-1.133926392f, -0.459472656f, -0.129577637f, -0.056533810f,
+-0.031814575f, -0.003723145f, -0.002899170f, -0.000289917f,
+0.000015259f, 0.000686646f, -0.003479004f, 0.011886597f,
+-0.026535034f, 0.105819702f, -0.044784546f, 0.711318970f,
+-1.127746582f, -0.431655884f, -0.134887695f, -0.051132202f,
+-0.031661987f, -0.003005981f, -0.002792358f, -0.000259399f,
+0.000015259f, 0.000747681f, -0.003479004f, 0.012939452f,
+-0.025085449f, 0.110946655f, -0.031082151f, 0.738372803f,
+-1.120223999f, -0.404083252f, -0.139450073f, -0.045837402f,
+-0.031387329f, -0.002334595f, -0.002685547f, -0.000244141f,
+0.000030518f, 0.000808716f, -0.003463745f, 0.014022826f,
+-0.023422241f, 0.115921021f, -0.016510010f, 0.765029907f,
+-1.111373901f, -0.376800537f, -0.143264771f, -0.040634155f,
+-0.031005858f, -0.001693726f, -0.002578735f, -0.000213623f,
+0.000030518f, 0.000885010f, -0.003417969f, 0.015121460f,
+-0.021575928f, 0.120697014f, -0.001068115f, 0.791213989f,
+-1.101211548f, -0.349868774f, -0.146362305f, -0.035552979f,
+-0.030532837f, -0.001098633f, -0.002456665f, -0.000198364f,
+0.000030518f, 0.000961304f, -0.003372192f, 0.016235352f,
+-0.019531250f, 0.125259399f, 0.015228271f, 0.816864014f,
+-1.089782715f, -0.323318481f, -0.148773193f, -0.030609131f,
+-0.029937742f, -0.000549316f, -0.002349854f, -0.000167847f,
+0.000030518f, 0.001037598f, -0.003280640f, 0.017349243f,
+-0.017257690f, 0.129562378f, 0.032379150f, 0.841949463f,
+-1.077117920f, -0.297210693f, -0.150497437f, -0.025817871f,
+-0.029281614f, -0.000030518f, -0.002243042f, -0.000152588f,
+0.000045776f, 0.001113892f, -0.003173828f, 0.018463135f,
+-0.014801024f, 0.133590698f, 0.050354004f, 0.866363525f,
+-1.063217163f, -0.271591187f, -0.151596069f, -0.021179199f,
+-0.028533936f, 0.000442505f, -0.002120972f, -0.000137329f,
+0.000045776f, 0.001205444f, -0.003051758f, 0.019577026f,
+-0.012115479f, 0.137298584f, 0.069168091f, 0.890090942f,
+-1.048156738f, -0.246505737f, -0.152069092f, -0.016708374f,
+-0.027725220f, 0.000869751f, -0.002014160f, -0.000122070f,
+0.000061035f, 0.001296997f, -0.002883911f, 0.020690918f,
+-0.009231566f, 0.140670776f, 0.088775635f, 0.913055420f,
+-1.031936646f, -0.221984863f, -0.151962280f, -0.012420653f,
+-0.026840210f, 0.001266479f, -0.001907349f, -0.000106812f,
+0.000061035f, 0.001388550f, -0.002700806f, 0.021789551f,
+-0.006134033f, 0.143676758f, 0.109161377f, 0.935195923f,
+-1.014617920f, -0.198059082f, -0.151306152f, -0.008316040f,
+-0.025909424f, 0.001617432f, -0.001785278f, -0.000106812f,
+0.000076294f, 0.001480103f, -0.002487183f, 0.022857666f,
+-0.002822876f, 0.146255493f, 0.130310059f, 0.956481934f,
+-0.996246338f, -0.174789429f, -0.150115967f, -0.004394531f,
+-0.024932859f, 0.001937866f, -0.001693726f, -0.000091553f,
+-0.001586914f, -0.023910521f, -0.148422241f, -0.976852417f,
+0.152206421f, 0.000686646f, -0.002227783f, 0.000076294f,

codecs/mp3/include/xinglmc.h

@@ -0,0 +1,166 @@
+/*____________________________________________________________________________
+   
+   FreeAmp - The Free MP3 Player
+   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.
+   
+   $Id$
+
+____________________________________________________________________________*/
+
+#ifndef INCLUDED_XINGLMC_H_
+#define INCLUDED_XINGLMC_H_
+
+/* system headers */
+#include <stdlib.h>
+#include <time.h>
+
+/* project headers */
+#include "config.h"
+
+#include "pmi.h"
+#include "pmo.h"
+#include "mutex.h"
+#include "event.h"
+#include "lmc.h"
+#include "thread.h"
+#include "mutex.h"
+#include "queue.h"
+#include "semaphore.h"
+
+extern    "C"
+{
+#include "mhead.h"
+#include "port.h"
+}
+
+#define BS_BUFBYTES 60000U
+#define PCM_BUFBYTES 60000U
+
+typedef struct
+{
+   int       (*decode_init) (MPEG_HEAD * h, int framebytes_arg,
+              int reduction_code, int transform_code,
+              int convert_code, int freq_limit);
+   void      (*decode_info) (DEC_INFO * info);
+             IN_OUT(*decode) (unsigned char *bs, short *pcm);
+}
+AUDIO;
+
+#define FRAMES_FLAG     0x0001
+#define BYTES_FLAG      0x0002
+#define TOC_FLAG        0x0004
+#define VBR_SCALE_FLAG  0x0008
+
+#define FRAMES_AND_BYTES (FRAMES_FLAG | BYTES_FLAG)
+
+// structure to receive extracted header
+// toc may be NULL
+typedef struct 
+{
+    int h_id;       // from MPEG header, 0=MPEG2, 1=MPEG1
+    int samprate;   // determined from MPEG header
+    int flags;      // from Xing header data
+    int frames;     // total bit stream frames from Xing header data
+    int bytes;      // total bit stream bytes from Xing header data
+    int vbr_scale;  // encoded vbr scale from Xing header data
+    unsigned char *toc;  // pointer to unsigned char toc_buffer[100]
+                         // may be NULL if toc not desired
+}   XHEADDATA;
+
+enum
+{
+   lmcError_MinimumError = 1000,
+   lmcError_DecodeFailed,
+   lmcError_AudioDecodeInitFailed,
+   lmcError_DecoderThreadFailed,
+   lmcError_PMIError,
+   lmcError_PMOError,
+   lmcError_MaximumError
+};
+
+class     XingLMC:public LogicalMediaConverter
+{
+
+   public:
+            XingLMC(FAContext *context);
+   virtual ~XingLMC();
+
+   virtual uint32 CalculateSongLength(const char *url);
+
+   virtual Error ChangePosition(int32 position);
+
+   virtual Error CanDecode();
+   virtual void  Clear();
+   virtual Error ExtractMediaInfo();
+
+   virtual void  SetPMI(PhysicalMediaInput *pmi) { m_pPmi = pmi; };
+   virtual void  SetPMO(PhysicalMediaOutput *pmo) { m_pPmo = pmo; };
+   virtual Error Prepare(PullBuffer *pInputBuffer, PullBuffer *&pOutBuffer);
+   virtual Error InitDecoder();
+
+   virtual Error SetEQData(float *);
+   virtual Error SetEQData(bool);
+
+   virtual vector<char *> *GetExtensions(void);
+   
+ private:
+
+   static void          DecodeWorkerThreadFunc(void *);
+   void                 DecodeWork();
+   Error                BeginRead(void *&pBuffer, unsigned int iBytesNeeded,
+                                  bool bBufferUp = true);
+   Error                BlockingBeginRead(void *&pBuffer, 
+                                          unsigned int iBytesNeeded);
+   Error                EndRead(size_t iBytesUsed);
+   Error                AdvanceBufferToNextFrame();
+   Error                GetHeadInfo();
+   Error                GetBitstreamStats(float &fTotalSeconds, float &fMsPerFrame,
+                                          int &iTotalFrames, int &iSampleRate, 
+                                          int &iLayer);
+
+   int                  GetXingHeader(XHEADDATA *X,  unsigned char *buf);
+   int                  SeekPoint(unsigned char TOC[100], int file_bytes, float percent);
+   int                  ExtractI4(unsigned char *buf);
+
+   PhysicalMediaInput  *m_pPmi;
+   PhysicalMediaOutput *m_pPmo;
+
+   int                  m_iMaxWriteSize;
+   int32                m_frameBytes, m_iBufferUpInterval, m_iBufferSize;
+   size_t               m_lFileSize;
+   MPEG_HEAD            m_sMpegHead;
+   int32                m_iBitRate;
+   bool                 m_bBufferingUp;
+   Thread              *m_decoderThread;
+
+   int32                m_frameCounter;
+   time_t               m_iBufferUpdate;
+   char                *m_szUrl;
+   const char          *m_szError;
+   AUDIO                m_audioMethods; 
+   XHEADDATA           *m_pXingHeader;
+   
+   // These vars are used for a nasty hack.
+   FILE                *m_fpFile;
+   char                *m_pLocalReadBuffer;
+};
+
+#endif /* _XINGLMC_H */
+
+
+
+

codecs/mp3/src/cdct.c

@@ -0,0 +1,349 @@
+/*____________________________________________________________________________
+	
+	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.
+	
+	$Id$
+____________________________________________________________________________*/
+
+/****  cdct.c  ***************************************************
+
+mod 5/16/95 first stage in 8 pt dct does not drop last sb mono
+
+
+MPEG audio decoder, dct
+portable C
+
+******************************************************************/
+
+#include <stdlib.h>
+#include <stdio.h>
+#include <float.h>
+#include <math.h>
+#include "L3.h"
+#include "mhead.h"
+
+#ifdef ASM_X86
+extern void fdct32_asm(float*a, float*b);
+extern void fdct32_dual_asm(float*a, float*b);
+#endif /* ASM_X86 */
+
+/*------------------------------------------------------------*/
+float *dct_coef_addr(MPEG *m)
+{
+   return m->cdct.coef32;
+}
+/*------------------------------------------------------------*/
+static void forward_bf(int m, int n, float x[], float f[], float coef[])
+{
+   int i, j, n2;
+   int p, q, p0, k;
+
+   p0 = 0;
+   n2 = n >> 1;
+   for (i = 0; i < m; i++, p0 += n)
+   {
+      k = 0;
+      p = p0;
+      q = p + n - 1;
+      for (j = 0; j < n2; j++, p++, q--, k++)
+      {
+	 f[p] = x[p] + x[q];
+	 f[n2 + p] = coef[k] * (x[p] - x[q]);
+      }
+   }
+}
+/*------------------------------------------------------------*/
+static void back_bf(int m, int n, float x[], float f[])
+{
+   int i, j, n2, n21;
+   int p, q, p0;
+
+   p0 = 0;
+   n2 = n >> 1;
+   n21 = n2 - 1;
+   for (i = 0; i < m; i++, p0 += n)
+   {
+      p = p0;
+      q = p0;
+      for (j = 0; j < n2; j++, p += 2, q++)
+	 f[p] = x[q];
+      p = p0 + 1;
+      for (j = 0; j < n21; j++, p += 2, q++)
+	 f[p] = x[q] + x[q + 1];
+      f[p] = x[q];
+   }
+}
+/*------------------------------------------------------------*/
+
+#ifdef  _EQUALIZER_ENABLE_
+extern float equalizer[32];
+extern int enableEQ;
+#endif
+
+void fdct32(MPEG *m, float x[], float c[])
+{
+#if (!defined(ASM_X86) && !defined(ASM_X86_OLD) || defined(_EQUALIZER_ENABLE_))
+   float a[32];			/* ping pong buffers */
+   float b[32];
+   int p, q;
+#endif
+
+   float *src = x;
+
+#ifdef  _EQUALIZER_ENABLE_
+   int i;
+   float b[32];
+   if (enableEQ) {
+       for(i=0; i<32; i++)
+	   b[i] = x[i] * equalizer[i];
+       src = b;
+   }
+#endif  /* _EQUALIZER_ENABLE_ */
+#undef  _EQUALIZER_ENABLE_
+
+#ifdef ASM_X86
+   fdct32_asm(src, c);
+#elif defined(ASM_X86_OLD)
+   asm_fdct32(src, c);
+#else
+/* special first stage */
+   for (p = 0, q = 31; p < 16; p++, q--)
+   {
+      a[p] = src[p] + src[q];
+      a[16 + p] = m->cdct.coef32[p] * (src[p] - src[q]);
+   }
+   forward_bf(2, 16, a, b, m->cdct.coef32 + 16);
+   forward_bf(4, 8, b, a, m->cdct.coef32 + 16 + 8);
+   forward_bf(8, 4, a, b, m->cdct.coef32 + 16 + 8 + 4);
+   forward_bf(16, 2, b, a, m->cdct.coef32 + 16 + 8 + 4 + 2);
+   back_bf(8, 4, a, b);
+   back_bf(4, 8, b, a);
+   back_bf(2, 16, a, b);
+   back_bf(1, 32, b, c);
+#endif
+}
+/*------------------------------------------------------------*/
+void fdct32_dual(MPEG *m, float x[], float c[])
+{
+#ifdef ASM_X86
+   fdct32_dual_asm(x, c);
+#else
+   float a[32];			/* ping pong buffers */
+   float b[32];
+   int p, pp, qq;
+
+/* special first stage for dual chan (interleaved x) */
+   pp = 0;
+   qq = 2 * 31;
+   for (p = 0; p < 16; p++, pp += 2, qq -= 2)
+   {
+      a[p] = x[pp] + x[qq];
+      a[16 + p] = m->cdct.coef32[p] * (x[pp] - x[qq]);
+   }
+   forward_bf(2, 16, a, b, m->cdct.coef32 + 16);
+   forward_bf(4, 8, b, a, m->cdct.coef32 + 16 + 8);
+   forward_bf(8, 4, a, b, m->cdct.coef32 + 16 + 8 + 4);
+   forward_bf(16, 2, b, a, m->cdct.coef32 + 16 + 8 + 4 + 2);
+   back_bf(8, 4, a, b);
+   back_bf(4, 8, b, a);
+   back_bf(2, 16, a, b);
+   back_bf(1, 32, b, c);
+#endif
+}
+/*---------------convert dual to mono------------------------------*/
+void fdct32_dual_mono(MPEG *m, float x[], float c[])
+{
+   float a[32];			/* ping pong buffers */
+   float b[32];
+   float t1, t2;
+   int p, pp, qq;
+
+/* special first stage  */
+   pp = 0;
+   qq = 2 * 31;
+   for (p = 0; p < 16; p++, pp += 2, qq -= 2)
+   {
+      t1 = 0.5F * (x[pp] + x[pp + 1]);
+      t2 = 0.5F * (x[qq] + x[qq + 1]);
+      a[p] = t1 + t2;
+      a[16 + p] = m->cdct.coef32[p] * (t1 - t2);
+   }
+   forward_bf(2, 16, a, b, m->cdct.coef32 + 16);
+   forward_bf(4, 8, b, a, m->cdct.coef32 + 16 + 8);
+   forward_bf(8, 4, a, b, m->cdct.coef32 + 16 + 8 + 4);
+   forward_bf(16, 2, b, a, m->cdct.coef32 + 16 + 8 + 4 + 2);
+   back_bf(8, 4, a, b);
+   back_bf(4, 8, b, a);
+   back_bf(2, 16, a, b);
+   back_bf(1, 32, b, c);
+}
+/*------------------------------------------------------------*/
+/*---------------- 16 pt fdct -------------------------------*/
+void fdct16(MPEG *m, float x[], float c[])
+{
+   float a[16];			/* ping pong buffers */
+   float b[16];
+   int p, q;
+
+/* special first stage (drop highest sb) */
+   a[0] = x[0];
+   a[8] = m->cdct.coef32[16] * x[0];
+   for (p = 1, q = 14; p < 8; p++, q--)
+   {
+      a[p] = x[p] + x[q];
+      a[8 + p] = m->cdct.coef32[16 + p] * (x[p] - x[q]);
+   }
+   forward_bf(2, 8, a, b, m->cdct.coef32 + 16 + 8);
+   forward_bf(4, 4, b, a, m->cdct.coef32 + 16 + 8 + 4);
+   forward_bf(8, 2, a, b, m->cdct.coef32 + 16 + 8 + 4 + 2);
+   back_bf(4, 4, b, a);
+   back_bf(2, 8, a, b);
+   back_bf(1, 16, b, c);
+}
+/*------------------------------------------------------------*/
+/*---------------- 16 pt fdct dual chan---------------------*/
+void fdct16_dual(MPEG *m, float x[], float c[])
+{
+   float a[16];			/* ping pong buffers */
+   float b[16];
+   int p, pp, qq;
+
+/* special first stage for interleaved input */
+   a[0] = x[0];
+   a[8] = m->cdct.coef32[16] * x[0];
+   pp = 2;
+   qq = 2 * 14;
+   for (p = 1; p < 8; p++, pp += 2, qq -= 2)
+   {
+      a[p] = x[pp] + x[qq];
+      a[8 + p] = m->cdct.coef32[16 + p] * (x[pp] - x[qq]);
+   }
+   forward_bf(2, 8, a, b, m->cdct.coef32 + 16 + 8);
+   forward_bf(4, 4, b, a, m->cdct.coef32 + 16 + 8 + 4);
+   forward_bf(8, 2, a, b, m->cdct.coef32 + 16 + 8 + 4 + 2);
+   back_bf(4, 4, b, a);
+   back_bf(2, 8, a, b);
+   back_bf(1, 16, b, c);
+}
+/*------------------------------------------------------------*/
+/*---------------- 16 pt fdct dual to mono-------------------*/
+void fdct16_dual_mono(MPEG *m, float x[], float c[])
+{
+   float a[16];			/* ping pong buffers */
+   float b[16];
+   float t1, t2;
+   int p, pp, qq;
+
+/* special first stage  */
+   a[0] = 0.5F * (x[0] + x[1]);
+   a[8] = m->cdct.coef32[16] * a[0];
+   pp = 2;
+   qq = 2 * 14;
+   for (p = 1; p < 8; p++, pp += 2, qq -= 2)
+   {
+      t1 = 0.5F * (x[pp] + x[pp + 1]);
+      t2 = 0.5F * (x[qq] + x[qq + 1]);
+      a[p] = t1 + t2;
+      a[8 + p] = m->cdct.coef32[16 + p] * (t1 - t2);
+   }
+   forward_bf(2, 8, a, b, m->cdct.coef32 + 16 + 8);
+   forward_bf(4, 4, b, a, m->cdct.coef32 + 16 + 8 + 4);
+   forward_bf(8, 2, a, b, m->cdct.coef32 + 16 + 8 + 4 + 2);
+   back_bf(4, 4, b, a);
+   back_bf(2, 8, a, b);
+   back_bf(1, 16, b, c);
+}
+/*------------------------------------------------------------*/
+/*---------------- 8 pt fdct -------------------------------*/
+void fdct8(MPEG *m, float x[], float c[])
+{
+   float a[8];			/* ping pong buffers */
+   float b[8];
+   int p, q;
+
+/* special first stage  */
+
+   b[0] = x[0] + x[7];
+   b[4] = m->cdct.coef32[16 + 8] * (x[0] - x[7]);
+   for (p = 1, q = 6; p < 4; p++, q--)
+   {
+      b[p] = x[p] + x[q];
+      b[4 + p] = m->cdct.coef32[16 + 8 + p] * (x[p] - x[q]);
+   }
+
+   forward_bf(2, 4, b, a, m->cdct.coef32 + 16 + 8 + 4);
+   forward_bf(4, 2, a, b, m->cdct.coef32 + 16 + 8 + 4 + 2);
+   back_bf(2, 4, b, a);
+   back_bf(1, 8, a, c);
+}
+/*------------------------------------------------------------*/
+/*---------------- 8 pt fdct dual chan---------------------*/
+void fdct8_dual(MPEG *m, float x[], float c[])
+{
+   float a[8];			/* ping pong buffers */
+   float b[8];
+   int p, pp, qq;
+
+/* special first stage for interleaved input */
+   b[0] = x[0] + x[14];
+   b[4] = m->cdct.coef32[16 + 8] * (x[0] - x[14]);
+   pp = 2;
+   qq = 2 * 6;
+   for (p = 1; p < 4; p++, pp += 2, qq -= 2)
+   {
+      b[p] = x[pp] + x[qq];
+      b[4 + p] = m->cdct.coef32[16 + 8 + p] * (x[pp] - x[qq]);
+   }
+   forward_bf(2, 4, b, a, m->cdct.coef32 + 16 + 8 + 4);
+   forward_bf(4, 2, a, b, m->cdct.coef32 + 16 + 8 + 4 + 2);
+   back_bf(2, 4, b, a);
+   back_bf(1, 8, a, c);
+}
+/*------------------------------------------------------------*/
+/*---------------- 8 pt fdct dual to mono---------------------*/
+void fdct8_dual_mono(MPEG *m, float x[], float c[])
+{
+   float a[8];			/* ping pong buffers */
+   float b[8];
+   float t1, t2;
+   int p, pp, qq;
+
+/* special first stage  */
+   t1 = 0.5F * (x[0] + x[1]);
+   t2 = 0.5F * (x[14] + x[15]);
+   b[0] = t1 + t2;
+   b[4] = m->cdct.coef32[16 + 8] * (t1 - t2);
+   pp = 2;
+   qq = 2 * 6;
+   for (p = 1; p < 4; p++, pp += 2, qq -= 2)
+   {
+      t1 = 0.5F * (x[pp] + x[pp + 1]);
+      t2 = 0.5F * (x[qq] + x[qq + 1]);
+      b[p] = t1 + t2;
+      b[4 + p] = m->cdct.coef32[16 + 8 + p] * (t1 - t2);
+   }
+   forward_bf(2, 4, b, a, m->cdct.coef32 + 16 + 8 + 4);
+   forward_bf(4, 2, a, b, m->cdct.coef32 + 16 + 8 + 4 + 2);
+   back_bf(2, 4, b, a);
+   back_bf(1, 8, a, c);
+}
+/*------------------------------------------------------------*/

codecs/mp3/src/csbt.c

@@ -0,0 +1,345 @@
+/*____________________________________________________________________________
+	
+	FreeAmp - The Free MP3 Player
+
+        MP3 Decoder originally Copyright (C) 1995-1997 Xing Technology
+        Corp.  http://www.xingtech.com
+
+	Portions Copyright (C) 1998 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.
+	
+	$Id$
+____________________________________________________________________________*/
+
+/****  csbt.c  ***************************************************
+
+MPEG audio decoder, dct and window
+portable C
+
+1/7/96 mod for Layer III
+  
+******************************************************************/
+
+#include <stdlib.h>
+#include <stdio.h>
+#include <float.h>
+#include <math.h>
+#include "L3.h"
+#include "mhead.h"
+
+void fdct32(MPEG *m, float *, float *);
+void fdct32_dual(MPEG *m, float *, float *);
+void fdct32_dual_mono(MPEG *m, float *, float *);
+void fdct16(MPEG *m, float *, float *);
+void fdct16_dual(MPEG *m, float *, float *);
+void fdct16_dual_mono(MPEG *m, float *, float *);
+void fdct8(MPEG *m, float *, float *);
+void fdct8_dual(MPEG *m, float *, float *);
+void fdct8_dual_mono(MPEG *m, float *, float *);
+
+void window(float *, int , short *pcm);
+void window_dual(MPEG *m, float *, int , short *pcm);
+void window16(MPEG *m, float *, int , short *pcm);
+void window16_dual(MPEG *m, float *, int , short *pcm);
+void window8(MPEG *m, float *, int , short *pcm);
+void window8_dual(MPEG *m, float *, int , short *pcm);
+float *dct_coef_addr(MPEG *m);
+
+/*-------------------------------------------------------------------------*/
+/* circular window buffers */
+
+/*======================================================================*/
+static void gencoef(MPEG *m)		/* gen coef for N=32 (31 coefs) */
+{
+   int p, n, i, k;
+   double t, pi;
+   float *coef32;
+
+   coef32 = dct_coef_addr(m);
+
+   pi = 4.0 * atan(1.0);
+   n = 16;
+   k = 0;
+   for (i = 0; i < 5; i++, n = n / 2)
+   {
+
+      for (p = 0; p < n; p++, k++)
+      {
+	 t = (pi / (4 * n)) * (2 * p + 1);
+	 coef32[k] = (float) (0.50 / cos(t));
+      }
+   }
+}
+/*------------------------------------------------------------*/
+void sbt_init(MPEG *m)
+{
+   int i;
+
+   if (m->csbt.first_pass)
+   {
+      gencoef(m);
+      m->csbt.first_pass = 0;
+   }
+
+/* clear window m->csbt.vbuf */
+   for (i = 0; i < 512; i++)
+   {
+      m->csbt.vbuf[i] = 0.0F;
+      m->csbt.vbuf2[i] = 0.0F;
+   }
+   m->csbt.vb2_ptr = m->csbt.vb_ptr = 0;
+
+}
+/*============================================================*/
+/*============================================================*/
+/*============================================================*/
+void sbt_mono(MPEG *m, float *sample, short *pcm, int n)
+{
+   int i;
+
+   for (i = 0; i < n; i++)
+   {
+      fdct32(m, sample, m->csbt.vbuf + m->csbt.vb_ptr);
+      window(m->csbt.vbuf, m->csbt.vb_ptr, pcm);
+      sample += 64;
+      m->csbt.vb_ptr = (m->csbt.vb_ptr - 32) & 511;
+      pcm += 32;
+   }
+
+}
+/*------------------------------------------------------------*/
+void sbt_dual(MPEG *m, float *sample, short *pcm, int n)
+{
+   int i;
+
+   for (i = 0; i < n; i++)
+   {
+      fdct32_dual(m,sample, m->csbt.vbuf + m->csbt.vb_ptr);
+      fdct32_dual(m,sample + 1, m->csbt.vbuf2 + m->csbt.vb_ptr);
+      window_dual(m,m->csbt.vbuf, m->csbt.vb_ptr, pcm);
+      window_dual(m,m->csbt.vbuf2, m->csbt.vb_ptr, pcm + 1);
+      sample += 64;
+      m->csbt.vb_ptr = (m->csbt.vb_ptr - 32) & 511;
+      pcm += 64;
+   }
+
+
+}
+/*------------------------------------------------------------*/
+/* convert dual to mono */
+void sbt_dual_mono(MPEG *m, float *sample, short *pcm, int n)
+{
+   int i;
+
+   for (i = 0; i < n; i++)
+   {
+      fdct32_dual_mono(m,sample, m->csbt.vbuf + m->csbt.vb_ptr);
+      window(m->csbt.vbuf, m->csbt.vb_ptr, pcm);
+      sample += 64;
+      m->csbt.vb_ptr = (m->csbt.vb_ptr - 32) & 511;
+      pcm += 32;
+   }
+
+}
+/*------------------------------------------------------------*/
+/* convert dual to left */
+void sbt_dual_left(MPEG *m, float *sample, short *pcm, int n)
+{
+   int i;
+
+   for (i = 0; i < n; i++)
+   {
+      fdct32_dual(m,sample, m->csbt.vbuf + m->csbt.vb_ptr);
+      window(m->csbt.vbuf, m->csbt.vb_ptr, pcm);
+      sample += 64;
+      m->csbt.vb_ptr = (m->csbt.vb_ptr - 32) & 511;
+      pcm += 32;
+   }
+}
+/*------------------------------------------------------------*/
+/* convert dual to right */
+void sbt_dual_right(MPEG *m, float *sample, short *pcm, int n)
+{
+   int i;
+
+   sample++;			/* point to right chan */
+   for (i = 0; i < n; i++)
+   {
+      fdct32_dual(m,sample, m->csbt.vbuf + m->csbt.vb_ptr);
+      window(m->csbt.vbuf, m->csbt.vb_ptr, pcm);
+      sample += 64;
+      m->csbt.vb_ptr = (m->csbt.vb_ptr - 32) & 511;
+      pcm += 32;
+   }
+}
+/*------------------------------------------------------------*/
+/*---------------- 16 pt sbt's  -------------------------------*/
+/*------------------------------------------------------------*/
+void sbt16_mono(MPEG *m, float *sample, short *pcm, int n)
+{
+   int i;
+
+   for (i = 0; i < n; i++)
+   {
+      fdct16(m,sample, m->csbt.vbuf + m->csbt.vb_ptr);
+      window16(m,m->csbt.vbuf, m->csbt.vb_ptr, pcm);
+      sample += 64;
+      m->csbt.vb_ptr = (m->csbt.vb_ptr - 16) & 255;
+      pcm += 16;
+   }
+
+
+}
+/*------------------------------------------------------------*/
+void sbt16_dual(MPEG *m, float *sample, short *pcm, int n)
+{
+   int i;
+
+   for (i = 0; i < n; i++)
+   {
+      fdct16_dual(m,sample, m->csbt.vbuf + m->csbt.vb_ptr);
+      fdct16_dual(m,sample + 1, m->csbt.vbuf2 + m->csbt.vb_ptr);
+      window16_dual(m,m->csbt.vbuf, m->csbt.vb_ptr, pcm);
+      window16_dual(m,m->csbt.vbuf2, m->csbt.vb_ptr, pcm + 1);
+      sample += 64;
+      m->csbt.vb_ptr = (m->csbt.vb_ptr - 16) & 255;
+      pcm += 32;
+   }
+}
+/*------------------------------------------------------------*/
+void sbt16_dual_mono(MPEG *m, float *sample, short *pcm, int n)
+{
+   int i;
+
+   for (i = 0; i < n; i++)
+   {
+      fdct16_dual_mono(m,sample, m->csbt.vbuf + m->csbt.vb_ptr);
+      window16(m,m->csbt.vbuf, m->csbt.vb_ptr, pcm);
+      sample += 64;
+      m->csbt.vb_ptr = (m->csbt.vb_ptr - 16) & 255;
+      pcm += 16;
+   }
+}
+/*------------------------------------------------------------*/
+void sbt16_dual_left(MPEG *m, float *sample, short *pcm, int n)
+{
+   int i;
+
+   for (i = 0; i < n; i++)
+   {
+      fdct16_dual(m,sample, m->csbt.vbuf + m->csbt.vb_ptr);
+      window16(m,m->csbt.vbuf, m->csbt.vb_ptr, pcm);
+      sample += 64;
+      m->csbt.vb_ptr = (m->csbt.vb_ptr - 16) & 255;
+      pcm += 16;
+   }
+}
+/*------------------------------------------------------------*/
+void sbt16_dual_right(MPEG *m, float *sample, short *pcm, int n)
+{
+   int i;
+
+   sample++;
+   for (i = 0; i < n; i++)
+   {
+      fdct16_dual(m,sample, m->csbt.vbuf + m->csbt.vb_ptr);
+      window16(m,m->csbt.vbuf, m->csbt.vb_ptr, pcm);
+      sample += 64;
+      m->csbt.vb_ptr = (m->csbt.vb_ptr - 16) & 255;
+      pcm += 16;
+   }
+}
+/*------------------------------------------------------------*/
+/*---------------- 8 pt sbt's  -------------------------------*/
+/*------------------------------------------------------------*/
+void sbt8_mono(MPEG *m, float *sample, short *pcm, int n)
+{
+   int i;
+
+   for (i = 0; i < n; i++)
+   {
+      fdct8(m,sample, m->csbt.vbuf + m->csbt.vb_ptr);
+      window8(m,m->csbt.vbuf, m->csbt.vb_ptr, pcm);
+      sample += 64;
+      m->csbt.vb_ptr = (m->csbt.vb_ptr - 8) & 127;
+      pcm += 8;
+   }
+
+}
+/*------------------------------------------------------------*/
+void sbt8_dual(MPEG *m, float *sample, short *pcm, int n)
+{
+   int i;
+
+   for (i = 0; i < n; i++)
+   {
+      fdct8_dual(m,sample, m->csbt.vbuf + m->csbt.vb_ptr);
+      fdct8_dual(m,sample + 1, m->csbt.vbuf2 + m->csbt.vb_ptr);
+      window8_dual(m,m->csbt.vbuf, m->csbt.vb_ptr, pcm);
+      window8_dual(m,m->csbt.vbuf2, m->csbt.vb_ptr, pcm + 1);
+      sample += 64;
+      m->csbt.vb_ptr = (m->csbt.vb_ptr - 8) & 127;
+      pcm += 16;
+   }
+}
+/*------------------------------------------------------------*/
+void sbt8_dual_mono(MPEG *m, float *sample, short *pcm, int n)
+{
+   int i;
+
+   for (i = 0; i < n; i++)
+   {
+      fdct8_dual_mono(m,sample, m->csbt.vbuf + m->csbt.vb_ptr);
+      window8(m,m->csbt.vbuf, m->csbt.vb_ptr, pcm);
+      sample += 64;
+      m->csbt.vb_ptr = (m->csbt.vb_ptr - 8) & 127;
+      pcm += 8;
+   }
+}
+/*------------------------------------------------------------*/
+void sbt8_dual_left(MPEG *m, float *sample, short *pcm, int n)
+{
+   int i;
+
+   for (i = 0; i < n; i++)
+   {
+      fdct8_dual(m,sample, m->csbt.vbuf + m->csbt.vb_ptr);
+      window8(m,m->csbt.vbuf, m->csbt.vb_ptr, pcm);
+      sample += 64;
+      m->csbt.vb_ptr = (m->csbt.vb_ptr - 8) & 127;
+      pcm += 8;
+   }
+}
+/*------------------------------------------------------------*/
+void sbt8_dual_right(MPEG *m, float *sample, short *pcm, int n)
+{
+   int i;
+
+   sample++;
+   for (i = 0; i < n; i++)
+   {
+      fdct8_dual(m,sample, m->csbt.vbuf + m->csbt.vb_ptr);
+      window8(m,m->csbt.vbuf, m->csbt.vb_ptr, pcm);
+      sample += 64;
+      m->csbt.vb_ptr = (m->csbt.vb_ptr - 8) & 127;
+      pcm += 8;
+   }
+}
+/*------------------------------------------------------------*/
+/*------------------------------------------------------------*/
+#include "csbtb.c"		/* 8 bit output */
+#include "csbtL3.c"		/* Layer III */
+/*------------------------------------------------------------*/

codecs/mp3/src/csbtL3.c

@@ -0,0 +1,316 @@
+/*____________________________________________________________________________
+	
+	FreeAmp - The Free MP3 Player
+
+        MP3 Decoder originally Copyright (C) 1995-1997 Xing Technology
+        Corp.  http://www.xingtech.com
+
+	Portions Copyright (C) 1998 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.
+	
+	$Id$
+____________________________________________________________________________*/
+
+/****  csbtL3.c  ***************************************************
+
+layer III
+
+  include to  csbt.c
+
+******************************************************************/
+/*============================================================*/
+/*============ Layer III =====================================*/
+/*============================================================*/
+void sbt_mono_L3(MPEG *m, float *sample, short *pcm, int ch)
+{
+   int i;
+
+   ch = 0;
+   for (i = 0; i < 18; i++)
+   {
+      fdct32(m,sample, m->csbt.vbuf + m->csbt.vb_ptr);
+      window(m->csbt.vbuf, m->csbt.vb_ptr, pcm);
+      sample += 32;
+      m->csbt.vb_ptr = (m->csbt.vb_ptr - 32) & 511;
+      pcm += 32;
+   }
+
+}
+/*------------------------------------------------------------*/
+void sbt_dual_L3(MPEG *m, float *sample, short *pcm, int ch)
+{
+   int i;
+
+   if (ch == 0)
+      for (i = 0; i < 18; i++)
+      {
+	 fdct32(m,sample, m->csbt.vbuf + m->csbt.vb_ptr);
+	 window_dual(m,m->csbt.vbuf, m->csbt.vb_ptr, pcm);
+	 sample += 32;
+	 m->csbt.vb_ptr = (m->csbt.vb_ptr - 32) & 511;
+	 pcm += 64;
+      }
+   else
+      for (i = 0; i < 18; i++)
+      {
+	 fdct32(m,sample, m->csbt.vbuf2 + m->csbt.vb2_ptr);
+	 window_dual(m,m->csbt.vbuf2, m->csbt.vb2_ptr, pcm + 1);
+	 sample += 32;
+	 m->csbt.vb2_ptr = (m->csbt.vb2_ptr - 32) & 511;
+	 pcm += 64;
+      }
+
+
+}
+/*------------------------------------------------------------*/
+/*------------------------------------------------------------*/
+/*---------------- 16 pt sbt's  -------------------------------*/
+/*------------------------------------------------------------*/
+void sbt16_mono_L3(MPEG *m, float *sample, short *pcm, int ch)
+{
+   int i;
+
+   ch = 0;
+   for (i = 0; i < 18; i++)
+   {
+      fdct16(m,sample, m->csbt.vbuf + m->csbt.vb_ptr);
+      window16(m,m->csbt.vbuf, m->csbt.vb_ptr, pcm);
+      sample += 32;
+      m->csbt.vb_ptr = (m->csbt.vb_ptr - 16) & 255;
+      pcm += 16;
+   }
+
+
+}
+/*------------------------------------------------------------*/
+void sbt16_dual_L3(MPEG *m, float *sample, short *pcm, int ch)
+{
+   int i;
+
+
+   if (ch == 0)
+   {
+      for (i = 0; i < 18; i++)
+      {
+	 fdct16(m,sample, m->csbt.vbuf + m->csbt.vb_ptr);
+	 window16_dual(m,m->csbt.vbuf, m->csbt.vb_ptr, pcm);
+	 sample += 32;
+	 m->csbt.vb_ptr = (m->csbt.vb_ptr - 16) & 255;
+	 pcm += 32;
+      }
+   }
+   else
+   {
+      for (i = 0; i < 18; i++)
+      {
+	 fdct16(m,sample, m->csbt.vbuf2 + m->csbt.vb2_ptr);
+	 window16_dual(m,m->csbt.vbuf2, m->csbt.vb2_ptr, pcm + 1);
+	 sample += 32;
+	 m->csbt.vb2_ptr = (m->csbt.vb2_ptr - 16) & 255;
+	 pcm += 32;
+      }
+   }
+
+}
+/*------------------------------------------------------------*/
+/*---------------- 8 pt sbt's  -------------------------------*/
+/*------------------------------------------------------------*/
+void sbt8_mono_L3(MPEG *m, float *sample, short *pcm, int ch)
+{
+   int i;
+
+   ch = 0;
+   for (i = 0; i < 18; i++)
+   {
+      fdct8(m,sample, m->csbt.vbuf + m->csbt.vb_ptr);
+      window8(m,m->csbt.vbuf, m->csbt.vb_ptr, pcm);
+      sample += 32;
+      m->csbt.vb_ptr = (m->csbt.vb_ptr - 8) & 127;
+      pcm += 8;
+   }
+
+}
+/*------------------------------------------------------------*/
+void sbt8_dual_L3(MPEG *m, float *sample, short *pcm, int ch)
+{
+   int i;
+
+   if (ch == 0)
+   {
+      for (i = 0; i < 18; i++)
+      {
+	 fdct8(m,sample, m->csbt.vbuf + m->csbt.vb_ptr);
+	 window8_dual(m,m->csbt.vbuf, m->csbt.vb_ptr, pcm);
+	 sample += 32;
+	 m->csbt.vb_ptr = (m->csbt.vb_ptr - 8) & 127;
+	 pcm += 16;
+      }
+   }
+   else
+   {
+      for (i = 0; i < 18; i++)
+      {
+	 fdct8(m,sample, m->csbt.vbuf2 + m->csbt.vb2_ptr);
+	 window8_dual(m,m->csbt.vbuf2, m->csbt.vb2_ptr, pcm + 1);
+	 sample += 32;
+	 m->csbt.vb2_ptr = (m->csbt.vb2_ptr - 8) & 127;
+	 pcm += 16;
+      }
+   }
+
+
+
+}
+/*------------------------------------------------------------*/
+/*------- 8 bit output ---------------------------------------*/
+/*------------------------------------------------------------*/
+void sbtB_mono_L3(MPEG *m, float *sample, unsigned char *pcm, int ch)
+{
+   int i;
+
+   ch = 0;
+   for (i = 0; i < 18; i++)
+   {
+      fdct32(m,sample, m->csbt.vbuf + m->csbt.vb_ptr);
+      windowB(m,m->csbt.vbuf, m->csbt.vb_ptr, pcm);
+      sample += 32;
+      m->csbt.vb_ptr = (m->csbt.vb_ptr - 32) & 511;
+      pcm += 32;
+   }
+
+}
+/*------------------------------------------------------------*/
+void sbtB_dual_L3(MPEG *m, float *sample, unsigned char *pcm, int ch)
+{
+   int i;
+
+   if (ch == 0)
+      for (i = 0; i < 18; i++)
+      {
+	 fdct32(m,sample, m->csbt.vbuf + m->csbt.vb_ptr);
+	 windowB_dual(m,m->csbt.vbuf, m->csbt.vb_ptr, pcm);
+	 sample += 32;
+	 m->csbt.vb_ptr = (m->csbt.vb_ptr - 32) & 511;
+	 pcm += 64;
+      }
+   else
+      for (i = 0; i < 18; i++)
+      {
+	 fdct32(m,sample, m->csbt.vbuf2 + m->csbt.vb2_ptr);
+	 windowB_dual(m,m->csbt.vbuf2, m->csbt.vb2_ptr, pcm + 1);
+	 sample += 32;
+	 m->csbt.vb2_ptr = (m->csbt.vb2_ptr - 32) & 511;
+	 pcm += 64;
+      }
+
+}
+/*------------------------------------------------------------*/
+/*------------------------------------------------------------*/
+/*---------------- 16 pt sbtB's  -------------------------------*/
+/*------------------------------------------------------------*/
+void sbtB16_mono_L3(MPEG *m, float *sample, unsigned char *pcm, int ch)
+{
+   int i;
+
+   ch = 0;
+   for (i = 0; i < 18; i++)
+   {
+      fdct16(m,sample, m->csbt.vbuf + m->csbt.vb_ptr);
+      windowB16(m,m->csbt.vbuf, m->csbt.vb_ptr, pcm);
+      sample += 32;
+      m->csbt.vb_ptr = (m->csbt.vb_ptr - 16) & 255;
+      pcm += 16;
+   }
+
+
+}
+/*------------------------------------------------------------*/
+void sbtB16_dual_L3(MPEG *m, float *sample, unsigned char *pcm, int ch)
+{
+   int i;
+
+   if (ch == 0)
+   {
+      for (i = 0; i < 18; i++)
+      {
+	 fdct16(m,sample, m->csbt.vbuf + m->csbt.vb_ptr);
+	 windowB16_dual(m,m->csbt.vbuf, m->csbt.vb_ptr, pcm);
+	 sample += 32;
+	 m->csbt.vb_ptr = (m->csbt.vb_ptr - 16) & 255;
+	 pcm += 32;
+      }
+   }
+   else
+   {
+      for (i = 0; i < 18; i++)
+      {
+	 fdct16(m,sample, m->csbt.vbuf2 + m->csbt.vb2_ptr);
+	 windowB16_dual(m,m->csbt.vbuf2, m->csbt.vb2_ptr, pcm + 1);
+	 sample += 32;
+	 m->csbt.vb2_ptr = (m->csbt.vb2_ptr - 16) & 255;
+	 pcm += 32;
+      }
+   }
+
+}
+/*------------------------------------------------------------*/
+/*---------------- 8 pt sbtB's  -------------------------------*/
+/*------------------------------------------------------------*/
+void sbtB8_mono_L3(MPEG *m, float *sample, unsigned char *pcm, int ch)
+{
+   int i;
+
+   ch = 0;
+   for (i = 0; i < 18; i++)
+   {
+      fdct8(m,sample, m->csbt.vbuf + m->csbt.vb_ptr);
+      windowB8(m,m->csbt.vbuf, m->csbt.vb_ptr, pcm);
+      sample += 32;
+      m->csbt.vb_ptr = (m->csbt.vb_ptr - 8) & 127;
+      pcm += 8;
+   }
+
+}
+/*------------------------------------------------------------*/
+void sbtB8_dual_L3(MPEG *m, float *sample, unsigned char *pcm, int ch)
+{
+   int i;
+
+   if (ch == 0)
+   {
+      for (i = 0; i < 18; i++)
+      {
+	 fdct8(m,sample, m->csbt.vbuf + m->csbt.vb_ptr);
+	 windowB8_dual(m,m->csbt.vbuf, m->csbt.vb_ptr, pcm);
+	 sample += 32;
+	 m->csbt.vb_ptr = (m->csbt.vb_ptr - 8) & 127;
+	 pcm += 16;
+      }
+   }
+   else
+   {
+      for (i = 0; i < 18; i++)
+      {
+	 fdct8(m,sample, m->csbt.vbuf2 + m->csbt.vb2_ptr);
+	 windowB8_dual(m,m->csbt.vbuf2, m->csbt.vb2_ptr, pcm + 1);
+	 sample += 32;
+	 m->csbt.vb2_ptr = (m->csbt.vb2_ptr - 8) & 127;
+	 pcm += 16;
+      }
+   }
+
+}
+/*------------------------------------------------------------*/

codecs/mp3/src/csbtb.c

@@ -0,0 +1,276 @@
+/*____________________________________________________________________________
+	
+	FreeAmp - The Free MP3 Player
+
+        MP3 Decoder originally Copyright (C) 1995-1997 Xing Technology
+        Corp.  http://www.xingtech.com
+
+	Portions Copyright (C) 1998 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.
+	
+	$Id$
+____________________________________________________________________________*/
+
+/****  csbtb.c  ***************************************************
+include to csbt.c
+
+MPEG audio decoder, dct and window - byte (8 pcm bit output)
+portable C
+
+******************************************************************/
+/*============================================================*/
+/*============================================================*/
+void windowB(MPEG *m, float *, int , unsigned char *pcm);
+void windowB_dual(MPEG *m, float *, int , unsigned char *pcm);
+void windowB16(MPEG *m, float *, int , unsigned char *pcm);
+void windowB16_dual(MPEG *m, float *, int , unsigned char *pcm);
+void windowB8(MPEG *m, float *, int , unsigned char *pcm);
+void windowB8_dual(MPEG *m, float *, int , unsigned char *pcm);
+
+/*============================================================*/
+void sbtB_mono(MPEG *m, float *sample, unsigned char *pcm, int n)
+{
+   int i;
+
+   for (i = 0; i < n; i++)
+   {
+      fdct32(m,sample, m->csbt.vbuf + m->csbt.vb_ptr);
+      windowB(m,m->csbt.vbuf, m->csbt.vb_ptr, pcm);
+      sample += 64;
+      m->csbt.vb_ptr = (m->csbt.vb_ptr - 32) & 511;
+      pcm += 32;
+   }
+
+}
+/*------------------------------------------------------------*/
+void sbtB_dual(MPEG *m, float *sample, unsigned char *pcm, int n)
+{
+   int i;
+
+   for (i = 0; i < n; i++)
+   {
+      fdct32_dual(m,sample, m->csbt.vbuf + m->csbt.vb_ptr);
+      fdct32_dual(m,sample + 1, m->csbt.vbuf2 + m->csbt.vb_ptr);
+      windowB_dual(m,m->csbt.vbuf, m->csbt.vb_ptr, pcm);
+      windowB_dual(m,m->csbt.vbuf2, m->csbt.vb_ptr, pcm + 1);
+      sample += 64;
+      m->csbt.vb_ptr = (m->csbt.vb_ptr - 32) & 511;
+      pcm += 64;
+   }
+
+
+}
+/*------------------------------------------------------------*/
+/* convert dual to mono */
+void sbtB_dual_mono(MPEG *m, float *sample, unsigned char *pcm, int n)
+{
+   int i;
+
+   for (i = 0; i < n; i++)
+   {
+      fdct32_dual_mono(m,sample, m->csbt.vbuf + m->csbt.vb_ptr);
+      windowB(m,m->csbt.vbuf, m->csbt.vb_ptr, pcm);
+      sample += 64;
+      m->csbt.vb_ptr = (m->csbt.vb_ptr - 32) & 511;
+      pcm += 32;
+   }
+
+}
+/*------------------------------------------------------------*/
+/* convert dual to left */
+void sbtB_dual_left(MPEG *m, float *sample, unsigned char *pcm, int n)
+{
+   int i;
+
+   for (i = 0; i < n; i++)
+   {
+      fdct32_dual(m,sample, m->csbt.vbuf + m->csbt.vb_ptr);
+      windowB(m,m->csbt.vbuf, m->csbt.vb_ptr, pcm);
+      sample += 64;
+      m->csbt.vb_ptr = (m->csbt.vb_ptr - 32) & 511;
+      pcm += 32;
+   }
+}
+/*------------------------------------------------------------*/
+/* convert dual to right */
+void sbtB_dual_right(MPEG *m, float *sample, unsigned char *pcm, int n)
+{
+   int i;
+
+   sample++;			/* point to right chan */
+   for (i = 0; i < n; i++)
+   {
+      fdct32_dual(m,sample, m->csbt.vbuf + m->csbt.vb_ptr);
+      windowB(m,m->csbt.vbuf, m->csbt.vb_ptr, pcm);
+      sample += 64;
+      m->csbt.vb_ptr = (m->csbt.vb_ptr - 32) & 511;
+      pcm += 32;
+   }
+}
+/*------------------------------------------------------------*/
+/*---------------- 16 pt sbt's  -------------------------------*/
+/*------------------------------------------------------------*/
+void sbtB16_mono(MPEG *m, float *sample, unsigned char *pcm, int n)
+{
+   int i;
+
+   for (i = 0; i < n; i++)
+   {
+      fdct16(m,sample, m->csbt.vbuf + m->csbt.vb_ptr);
+      windowB16(m,m->csbt.vbuf, m->csbt.vb_ptr, pcm);
+      sample += 64;
+      m->csbt.vb_ptr = (m->csbt.vb_ptr - 16) & 255;
+      pcm += 16;
+   }
+
+
+}
+/*------------------------------------------------------------*/
+void sbtB16_dual(MPEG *m, float *sample, unsigned char *pcm, int n)
+{
+   int i;
+
+   for (i = 0; i < n; i++)
+   {
+      fdct16_dual(m,sample, m->csbt.vbuf + m->csbt.vb_ptr);
+      fdct16_dual(m,sample + 1, m->csbt.vbuf2 + m->csbt.vb_ptr);
+      windowB16_dual(m,m->csbt.vbuf, m->csbt.vb_ptr, pcm);
+      windowB16_dual(m,m->csbt.vbuf2, m->csbt.vb_ptr, pcm + 1);
+      sample += 64;
+      m->csbt.vb_ptr = (m->csbt.vb_ptr - 16) & 255;
+      pcm += 32;
+   }
+}
+/*------------------------------------------------------------*/
+void sbtB16_dual_mono(MPEG *m, float *sample, unsigned char *pcm, int n)
+{
+   int i;
+
+   for (i = 0; i < n; i++)
+   {
+      fdct16_dual_mono(m,sample, m->csbt.vbuf + m->csbt.vb_ptr);
+      windowB16(m,m->csbt.vbuf, m->csbt.vb_ptr, pcm);
+      sample += 64;
+      m->csbt.vb_ptr = (m->csbt.vb_ptr - 16) & 255;
+      pcm += 16;
+   }
+}
+/*------------------------------------------------------------*/
+void sbtB16_dual_left(MPEG *m, float *sample, unsigned char *pcm, int n)
+{
+   int i;
+
+   for (i = 0; i < n; i++)
+   {
+      fdct16_dual(m,sample, m->csbt.vbuf + m->csbt.vb_ptr);
+      windowB16(m,m->csbt.vbuf, m->csbt.vb_ptr, pcm);
+      sample += 64;
+      m->csbt.vb_ptr = (m->csbt.vb_ptr - 16) & 255;
+      pcm += 16;
+   }
+}
+/*------------------------------------------------------------*/
+void sbtB16_dual_right(MPEG *m, float *sample, unsigned char *pcm, int n)
+{
+   int i;
+
+   sample++;
+   for (i = 0; i < n; i++)
+   {
+      fdct16_dual(m,sample, m->csbt.vbuf + m->csbt.vb_ptr);
+      windowB16(m,m->csbt.vbuf, m->csbt.vb_ptr, pcm);
+      sample += 64;
+      m->csbt.vb_ptr = (m->csbt.vb_ptr - 16) & 255;
+      pcm += 16;
+   }
+}
+/*------------------------------------------------------------*/
+/*---------------- 8 pt sbt's  -------------------------------*/
+/*------------------------------------------------------------*/
+void sbtB8_mono(MPEG *m, float *sample, unsigned char *pcm, int n)
+{
+   int i;
+
+   for (i = 0; i < n; i++)
+   {
+      fdct8(m,sample, m->csbt.vbuf + m->csbt.vb_ptr);
+      windowB8(m,m->csbt.vbuf, m->csbt.vb_ptr, pcm);
+      sample += 64;
+      m->csbt.vb_ptr = (m->csbt.vb_ptr - 8) & 127;
+      pcm += 8;
+   }
+
+}
+/*------------------------------------------------------------*/
+void sbtB8_dual(MPEG *m, float *sample, unsigned char *pcm, int n)
+{
+   int i;
+
+   for (i = 0; i < n; i++)
+   {
+      fdct8_dual(m,sample, m->csbt.vbuf + m->csbt.vb_ptr);
+      fdct8_dual(m,sample + 1, m->csbt.vbuf2 + m->csbt.vb_ptr);
+      windowB8_dual(m,m->csbt.vbuf, m->csbt.vb_ptr, pcm);
+      windowB8_dual(m,m->csbt.vbuf2, m->csbt.vb_ptr, pcm + 1);
+      sample += 64;
+      m->csbt.vb_ptr = (m->csbt.vb_ptr - 8) & 127;
+      pcm += 16;
+   }
+}
+/*------------------------------------------------------------*/
+void sbtB8_dual_mono(MPEG *m, float *sample, unsigned char *pcm, int n)
+{
+   int i;
+
+   for (i = 0; i < n; i++)
+   {
+      fdct8_dual_mono(m,sample, m->csbt.vbuf + m->csbt.vb_ptr);
+      windowB8(m,m->csbt.vbuf, m->csbt.vb_ptr, pcm);
+      sample += 64;
+      m->csbt.vb_ptr = (m->csbt.vb_ptr - 8) & 127;
+      pcm += 8;
+   }
+}
+/*------------------------------------------------------------*/
+void sbtB8_dual_left(MPEG *m, float *sample, unsigned char *pcm, int n)
+{
+   int i;
+
+   for (i = 0; i < n; i++)
+   {
+      fdct8_dual(m,sample, m->csbt.vbuf + m->csbt.vb_ptr);
+      windowB8(m,m->csbt.vbuf, m->csbt.vb_ptr, pcm);
+      sample += 64;
+      m->csbt.vb_ptr = (m->csbt.vb_ptr - 8) & 127;
+      pcm += 8;
+   }
+}
+/*------------------------------------------------------------*/
+void sbtB8_dual_right(MPEG *m, float *sample, unsigned char *pcm, int n)
+{
+   int i;
+
+   sample++;
+   for (i = 0; i < n; i++)
+   {
+      fdct8_dual(m,sample, m->csbt.vbuf + m->csbt.vb_ptr);
+      windowB8(m,m->csbt.vbuf, m->csbt.vb_ptr, pcm);
+      sample += 64;
+      m->csbt.vb_ptr = (m->csbt.vb_ptr - 8) & 127;
+      pcm += 8;
+   }
+}
+/*------------------------------------------------------------*/

codecs/mp3/src/cup.c

@@ -0,0 +1,500 @@
+/*____________________________________________________________________________
+	
+	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.
+	
+	$Id$
+____________________________________________________________________________*/
+
+/****  cup.c  ***************************************************
+
+MPEG audio decoder Layer I/II  mpeg1 and mpeg2
+should be portable ANSI C, should be endian independent
+
+
+mod  2/21/95 2/21/95  add bit skip, sb limiting
+
+mods 11/15/95 for Layer I
+
+******************************************************************/
+/******************************************************************
+
+       MPEG audio software decoder portable ANSI c.
+       Decodes all Layer I/II to 16 bit linear pcm.
+       Optional stereo to mono conversion.  Optional
+       output sample rate conversion to half or quarter of
+       native mpeg rate. dec8.c adds oupuut conversion features.
+
+-------------------------------------
+int audio_decode_init(MPEG *m, MPEG_HEAD *h, int framebytes_arg,
+         int reduction_code, int transform_code, int convert_code,
+         int freq_limit)
+
+initilize decoder:
+       return 0 = fail, not 0 = success
+
+MPEG *m			input, mpeg structure for multiple streams
+MPEG_HEAD *h    input, mpeg header info (returned by call to head_info)
+framebytes      input, mpeg frame size (returned by call to head_info)
+reduction_code  input, sample rate reduction code
+                    0 = full rate
+                    1 = half rate
+                    2 = quarter rate
+
+transform_code  input, ignored
+convert_code    input, channel conversion
+                  convert_code:  0 = two chan output
+                                 1 = convert two chan to mono
+                                 2 = convert two chan to left chan
+                                 3 = convert two chan to right chan
+freq_limit      input, limits bandwidth of pcm output to specified
+                frequency.  Special use. Set to 24000 for normal use.
+
+
+---------------------------------
+void audio_decode_info( MPEG *m, DEC_INFO *info)
+
+MPEG *m			input, mpeg structure for multiple streams
+information return:
+          Call after audio_decode_init.  See mhead.h for
+          information returned in DEC_INFO structure.
+
+
+---------------------------------
+IN_OUT audio_decode(unsigned char *bs, void *pcmbuf)
+
+decode one mpeg audio frame:
+bs        input, mpeg bitstream, must start with
+          sync word.  Caution: may read up to 3 bytes
+          beyond end of frame.
+pcmbuf    output, pcm samples.
+
+IN_OUT structure returns:
+          Number bytes conceptually removed from mpeg bitstream.
+          Returns 0 if sync loss.
+          Number bytes of pcm output.
+
+*******************************************************************/
+
+
+#include <stdlib.h>
+#include <stdio.h>
+#include <float.h>
+#include <math.h>
+#include "L3.h"
+#include "mhead.h"		/* mpeg header structure */
+
+
+#ifdef _MSC_VER
+#pragma warning(disable: 4709)
+#endif
+
+
+/*-------------------------------------------------------
+NOTE:  Decoder may read up to three bytes beyond end of
+frame.  Calling application must ensure that this does
+not cause a memory access violation (protection fault)
+---------------------------------------------------------*/
+
+/*====================================================================*/
+/*----------------*/
+
+/* Read Only */
+static int look_joint[16] =
+{				/* lookup stereo sb's by mode+ext */
+   64, 64, 64, 64,		/* stereo */
+   2 * 4, 2 * 8, 2 * 12, 2 * 16,	/* joint */
+   64, 64, 64, 64,		/* dual */
+   32, 32, 32, 32,		/* mono */
+};	/* Okay to be global */
+
+/* Read Only */
+static int bat_bit_master[] =
+{
+   0, 5, 7, 9, 10, 12, 15, 18, 21, 24, 27, 30, 33, 36, 39, 42, 45, 48};
+
+
+void sbt_mono(MPEG *m, float *sample, short *pcm, int n);
+void sbt_dual(MPEG *m, float *sample, short *pcm, int n);
+
+
+IN_OUT L2audio_decode(void *mv, unsigned char *bs, signed short *pcm);
+
+/*======================================================================*/
+/*======================================================================*/
+
+/*------------- initialize bit getter -------------*/
+static void load_init(MPEG *m, unsigned char *buf)
+{
+   m->cup.bs_ptr = buf;
+   m->cup.bits = 0;
+   m->cup.bitbuf = 0;
+}
+/*------------- get n bits from bitstream -------------*/
+static long load(MPEG *m, int n)
+{
+   unsigned long x;
+
+   if (m->cup.bits < n)
+   {				/* refill bit buf if necessary */
+      while (m->cup.bits <= 24)
+      {
+	 m->cup.bitbuf = (m->cup.bitbuf << 8) | *m->cup.bs_ptr++;
+	 m->cup.bits += 8;
+      }
+   }
+   m->cup.bits -= n;
+   x = m->cup.bitbuf >> m->cup.bits;
+   m->cup.bitbuf -= x << m->cup.bits;
+   return x;
+}
+/*------------- skip over n bits in bitstream -------------*/
+static void skip(MPEG *m, int n)
+{
+   int k;
+
+   if (m->cup.bits < n)
+   {
+      n -= m->cup.bits;
+      k = n >> 3;
+/*--- bytes = n/8 --*/
+      m->cup.bs_ptr += k;
+      n -= k << 3;
+      m->cup.bitbuf = *m->cup.bs_ptr++;
+      m->cup.bits = 8;
+   }
+   m->cup.bits -= n;
+   m->cup.bitbuf -= (m->cup.bitbuf >> m->cup.bits) << m->cup.bits;
+}
+/*--------------------------------------------------------------*/
+#define mac_load_check(n) if( m->cup.bits < (n) ) {                           \
+          while( m->cup.bits <= 24 ) {               \
+             m->cup.bitbuf = (m->cup.bitbuf << 8) | *m->cup.bs_ptr++;  \
+             m->cup.bits += 8;                       \
+          }                                   \
+   }
+/*--------------------------------------------------------------*/
+#define mac_load(n) ( m->cup.bits -= n,                    \
+         m->cup.bitval = m->cup.bitbuf >> m->cup.bits,      \
+         m->cup.bitbuf -= m->cup.bitval << m->cup.bits,     \
+         m->cup.bitval )
+/*======================================================================*/
+static void unpack_ba(MPEG *m)
+{
+   int i, j, k;
+   /* Read Only */
+   static int nbit[4] =
+   {4, 4, 3, 2};
+   int nstereo;
+
+   m->cup.bit_skip = 0;
+   nstereo = m->cup.stereo_sb;
+   k = 0;
+   for (i = 0; i < 4; i++)
+   {
+      for (j = 0; j < m->cup.nbat[i]; j++, k++)
+      {
+	 mac_load_check(4);
+	 m->cup.ballo[k] = m->cup.samp_dispatch[k] = m->cup.bat[i][mac_load(nbit[i])];
+	 if (k >= m->cup.nsb_limit)
+	    m->cup.bit_skip += bat_bit_master[m->cup.samp_dispatch[k]];
+	 m->cup.c_value[k] = m->cup.look_c_value[m->cup.samp_dispatch[k]];
+	 if (--nstereo < 0)
+	 {
+	    m->cup.ballo[k + 1] = m->cup.ballo[k];
+	    m->cup.samp_dispatch[k] += 18;	/* flag as joint */
+	    m->cup.samp_dispatch[k + 1] = m->cup.samp_dispatch[k];	/* flag for sf */
+	    m->cup.c_value[k + 1] = m->cup.c_value[k];
+	    k++;
+	    j++;
+	 }
+      }
+   }
+   m->cup.samp_dispatch[m->cup.nsb_limit] = 37;	/* terminate the dispatcher with skip */
+   m->cup.samp_dispatch[k] = 36;	/* terminate the dispatcher */
+
+}
+/*-------------------------------------------------------------------------*/
+static void unpack_sfs(MPEG *m)	/* unpack scale factor selectors */
+{
+   int i;
+   for (i = 0; i < m->cup.max_sb; i++)
+   {
+      mac_load_check(2);
+      if (m->cup.ballo[i])
+	 m->cup.sf_dispatch[i] = mac_load(2);
+      else
+	 m->cup.sf_dispatch[i] = 4;	/* no allo */
+   }
+   m->cup.sf_dispatch[i] = 5;		/* terminate dispatcher */
+}
+/*-------------------------------------------------------------------------*/
+static void unpack_sf(MPEG *m)		/* unpack scale factor */
+{				/* combine dequant and scale factors */
+   int i;
+
+   i = -1;
+ dispatch:switch (m->cup.sf_dispatch[++i])
+   {
+      case 0:			/* 3 factors 012 */
+	 mac_load_check(18);
+	 m->cup.cs_factor[0][i] = m->cup.c_value[i] * m->cup.sf_table[mac_load(6)];
+	 m->cup.cs_factor[1][i] = m->cup.c_value[i] * m->cup.sf_table[mac_load(6)];
+	 m->cup.cs_factor[2][i] = m->cup.c_value[i] * m->cup.sf_table[mac_load(6)];
+	 goto dispatch;
+      case 1:			/* 2 factors 002 */
+	 mac_load_check(12);
+	 m->cup.cs_factor[1][i] = m->cup.cs_factor[0][i] = m->cup.c_value[i] * m->cup.sf_table[mac_load(6)];
+	 m->cup.cs_factor[2][i] = m->cup.c_value[i] * m->cup.sf_table[mac_load(6)];
+	 goto dispatch;
+      case 2:			/* 1 factor 000 */
+	 mac_load_check(6);
+	 m->cup.cs_factor[2][i] = m->cup.cs_factor[1][i] = m->cup.cs_factor[0][i] =
+	    m->cup.c_value[i] * m->cup.sf_table[mac_load(6)];
+	 goto dispatch;
+      case 3:			/* 2 factors 022 */
+	 mac_load_check(12);
+	 m->cup.cs_factor[0][i] = m->cup.c_value[i] * m->cup.sf_table[mac_load(6)];
+	 m->cup.cs_factor[2][i] = m->cup.cs_factor[1][i] = m->cup.c_value[i] * m->cup.sf_table[mac_load(6)];
+	 goto dispatch;
+      case 4:			/* no allo */
+/*-- m->cup.cs_factor[2][i] = m->cup.cs_factor[1][i] = m->cup.cs_factor[0][i] = 0.0;  --*/
+	 goto dispatch;
+      case 5:			/* all done */
+	 ;
+   }				/* end switch */
+}
+/*-------------------------------------------------------------------------*/
+#define UNPACK_N(n) s[k]     =  m->cup.cs_factor[i][k]*(load(m,n)-((1 << (n-1)) -1));   \
+    s[k+64]  =  m->cup.cs_factor[i][k]*(load(m,n)-((1 << (n-1)) -1));   \
+    s[k+128] =  m->cup.cs_factor[i][k]*(load(m,n)-((1 << (n-1)) -1));   \
+    goto dispatch;
+#define UNPACK_N2(n) mac_load_check(3*n);                                         \
+    s[k]     =  m->cup.cs_factor[i][k]*(mac_load(n)-((1 << (n-1)) -1));   \
+    s[k+64]  =  m->cup.cs_factor[i][k]*(mac_load(n)-((1 << (n-1)) -1));   \
+    s[k+128] =  m->cup.cs_factor[i][k]*(mac_load(n)-((1 << (n-1)) -1));   \
+    goto dispatch;
+#define UNPACK_N3(n) mac_load_check(2*n);                                         \
+    s[k]     =  m->cup.cs_factor[i][k]*(mac_load(n)-((1 << (n-1)) -1));   \
+    s[k+64]  =  m->cup.cs_factor[i][k]*(mac_load(n)-((1 << (n-1)) -1));   \
+    mac_load_check(n);                                           \
+    s[k+128] =  m->cup.cs_factor[i][k]*(mac_load(n)-((1 << (n-1)) -1));   \
+    goto dispatch;
+#define UNPACKJ_N(n) tmp        =  (load(m,n)-((1 << (n-1)) -1));                \
+    s[k]       =  m->cup.cs_factor[i][k]*tmp;                       \
+    s[k+1]     =  m->cup.cs_factor[i][k+1]*tmp;                     \
+    tmp        =  (load(m,n)-((1 << (n-1)) -1));                 \
+    s[k+64]    =  m->cup.cs_factor[i][k]*tmp;                       \
+    s[k+64+1]  =  m->cup.cs_factor[i][k+1]*tmp;                     \
+    tmp        =  (load(m,n)-((1 << (n-1)) -1));                 \
+    s[k+128]   =  m->cup.cs_factor[i][k]*tmp;                       \
+    s[k+128+1] =  m->cup.cs_factor[i][k+1]*tmp;                     \
+    k++;       /* skip right chan dispatch */                \
+    goto dispatch;
+/*-------------------------------------------------------------------------*/
+static void unpack_samp(MPEG *m)	/* unpack samples */
+{
+   int i, j, k;
+   float *s;
+   int n;
+   long tmp;
+
+   s = m->cup.sample;
+   for (i = 0; i < 3; i++)
+   {				/* 3 groups of scale factors */
+      for (j = 0; j < 4; j++)
+      {
+	 k = -1;
+       dispatch:switch (m->cup.samp_dispatch[++k])
+	 {
+	    case 0:
+	       s[k + 128] = s[k + 64] = s[k] = 0.0F;
+	       goto dispatch;
+	    case 1:		/* 3 levels grouped 5 bits */
+	       mac_load_check(5);
+	       n = mac_load(5);
+	       s[k] = m->cup.cs_factor[i][k] * m->cup.group3_table[n][0];
+	       s[k + 64] = m->cup.cs_factor[i][k] * m->cup.group3_table[n][1];
+	       s[k + 128] = m->cup.cs_factor[i][k] * m->cup.group3_table[n][2];
+	       goto dispatch;
+	    case 2:		/* 5 levels grouped 7 bits */
+	       mac_load_check(7);
+	       n = mac_load(7);
+	       s[k] = m->cup.cs_factor[i][k] * m->cup.group5_table[n][0];
+	       s[k + 64] = m->cup.cs_factor[i][k] * m->cup.group5_table[n][1];
+	       s[k + 128] = m->cup.cs_factor[i][k] * m->cup.group5_table[n][2];
+	       goto dispatch;
+	    case 3:
+	       UNPACK_N2(3)	/* 7 levels */
+	    case 4:		/* 9 levels grouped 10 bits */
+	       mac_load_check(10);
+	       n = mac_load(10);
+	       s[k] = m->cup.cs_factor[i][k] * m->cup.group9_table[n][0];
+	       s[k + 64] = m->cup.cs_factor[i][k] * m->cup.group9_table[n][1];
+	       s[k + 128] = m->cup.cs_factor[i][k] * m->cup.group9_table[n][2];
+	       goto dispatch;
+	    case 5:
+	       UNPACK_N2(4)	/* 15 levels */
+	    case 6:
+	       UNPACK_N2(5)	/* 31 levels */
+	    case 7:
+	       UNPACK_N2(6)	/* 63 levels */
+	    case 8:
+	       UNPACK_N2(7)	/* 127 levels */
+	    case 9:
+	       UNPACK_N2(8)	/* 255 levels */
+	    case 10:
+	       UNPACK_N3(9)	/* 511 levels */
+	    case 11:
+	       UNPACK_N3(10)	/* 1023 levels */
+	    case 12:
+	       UNPACK_N3(11)	/* 2047 levels */
+	    case 13:
+	       UNPACK_N3(12)	/* 4095 levels */
+	    case 14:
+	       UNPACK_N(13)	/* 8191 levels */
+	    case 15:
+	       UNPACK_N(14)	/* 16383 levels */
+	    case 16:
+	       UNPACK_N(15)	/* 32767 levels */
+	    case 17:
+	       UNPACK_N(16)	/* 65535 levels */
+/* -- joint ---- */
+	    case 18 + 0:
+	       s[k + 128 + 1] = s[k + 128] = s[k + 64 + 1] = s[k + 64] = s[k + 1] = s[k] = 0.0F;
+	       k++;		/* skip right chan dispatch */
+	       goto dispatch;
+	    case 18 + 1:	/* 3 levels grouped 5 bits */
+	       n = load(m,5);
+	       s[k] = m->cup.cs_factor[i][k] * m->cup.group3_table[n][0];
+	       s[k + 1] = m->cup.cs_factor[i][k + 1] * m->cup.group3_table[n][0];
+	       s[k + 64] = m->cup.cs_factor[i][k] * m->cup.group3_table[n][1];
+	       s[k + 64 + 1] = m->cup.cs_factor[i][k + 1] * m->cup.group3_table[n][1];
+	       s[k + 128] = m->cup.cs_factor[i][k] * m->cup.group3_table[n][2];
+	       s[k + 128 + 1] = m->cup.cs_factor[i][k + 1] * m->cup.group3_table[n][2];
+	       k++;		/* skip right chan dispatch */
+	       goto dispatch;
+	    case 18 + 2:	/* 5 levels grouped 7 bits */
+	       n = load(m,7);
+	       s[k] = m->cup.cs_factor[i][k] * m->cup.group5_table[n][0];
+	       s[k + 1] = m->cup.cs_factor[i][k + 1] * m->cup.group5_table[n][0];
+	       s[k + 64] = m->cup.cs_factor[i][k] * m->cup.group5_table[n][1];
+	       s[k + 64 + 1] = m->cup.cs_factor[i][k + 1] * m->cup.group5_table[n][1];
+	       s[k + 128] = m->cup.cs_factor[i][k] * m->cup.group5_table[n][2];
+	       s[k + 128 + 1] = m->cup.cs_factor[i][k + 1] * m->cup.group5_table[n][2];
+	       k++;		/* skip right chan dispatch */
+	       goto dispatch;
+	    case 18 + 3:
+	       UNPACKJ_N(3)	/* 7 levels */
+	    case 18 + 4:	/* 9 levels grouped 10 bits */
+	       n = load(m,10);
+	       s[k] = m->cup.cs_factor[i][k] * m->cup.group9_table[n][0];
+	       s[k + 1] = m->cup.cs_factor[i][k + 1] * m->cup.group9_table[n][0];
+	       s[k + 64] = m->cup.cs_factor[i][k] * m->cup.group9_table[n][1];
+	       s[k + 64 + 1] = m->cup.cs_factor[i][k + 1] * m->cup.group9_table[n][1];
+	       s[k + 128] = m->cup.cs_factor[i][k] * m->cup.group9_table[n][2];
+	       s[k + 128 + 1] = m->cup.cs_factor[i][k + 1] * m->cup.group9_table[n][2];
+	       k++;		/* skip right chan dispatch */
+	       goto dispatch;
+	    case 18 + 5:
+	       UNPACKJ_N(4)	/* 15 levels */
+	    case 18 + 6:
+	       UNPACKJ_N(5)	/* 31 levels */
+	    case 18 + 7:
+	       UNPACKJ_N(6)	/* 63 levels */
+	    case 18 + 8:
+	       UNPACKJ_N(7)	/* 127 levels */
+	    case 18 + 9:
+	       UNPACKJ_N(8)	/* 255 levels */
+	    case 18 + 10:
+	       UNPACKJ_N(9)	/* 511 levels */
+	    case 18 + 11:
+	       UNPACKJ_N(10)	/* 1023 levels */
+	    case 18 + 12:
+	       UNPACKJ_N(11)	/* 2047 levels */
+	    case 18 + 13:
+	       UNPACKJ_N(12)	/* 4095 levels */
+	    case 18 + 14:
+	       UNPACKJ_N(13)	/* 8191 levels */
+	    case 18 + 15:
+	       UNPACKJ_N(14)	/* 16383 levels */
+	    case 18 + 16:
+	       UNPACKJ_N(15)	/* 32767 levels */
+	    case 18 + 17:
+	       UNPACKJ_N(16)	/* 65535 levels */
+/* -- end of dispatch -- */
+	    case 37:
+	       skip(m, m->cup.bit_skip);
+	    case 36:
+	       s += 3 * 64;
+	 }			/* end switch */
+      }				/* end j loop */
+   }				/* end i loop */
+
+
+}
+/*-------------------------------------------------------------------------*/
+IN_OUT audio_decode(MPEG *m, unsigned char *bs, signed short *pcm)
+{
+   return m->cup.audio_decode_routine(m, bs, pcm);
+}
+/*-------------------------------------------------------------------------*/
+IN_OUT L2audio_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);
+   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_ba(m);			/* unpack bit allocation */
+   unpack_sfs(m);		/* unpack scale factor selectors */
+   unpack_sf(m);			/* unpack scale factor */
+   unpack_samp(m);		/* unpack samples */
+
+   m->cup.sbt(m, m->cup.sample, pcm, 36);
+/*-----------*/
+   in_out.in_bytes = m->cup.framebytes + m->cup.pad;
+   in_out.out_bytes = m->cup.outbytes;
+
+   return in_out;
+}
+/*-------------------------------------------------------------------------*/
+#include "cupini.c"		/* initialization */
+#include "cupL1.c"		/* Layer I */
+/*-------------------------------------------------------------------------*/

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;
+}
+/*---------------------------------------------------------*/

codecs/mp3/src/cupini.c

@@ -0,0 +1,421 @@
+/*____________________________________________________________________________
+	
+	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.
+	
+	$Id$
+____________________________________________________________________________*/
+
+/*=========================================================
+ initialization for cup.c - include to cup.c
+ mpeg audio decoder portable "c"
+
+mod 8/6/96 add 8 bit output
+
+mod 5/10/95 add quick (low precision) window
+
+mod 5/16/95 sb limit for reduced samprate output
+            changed from 94% to 100% of Nyquist sb
+
+mod 11/15/95 for Layer I
+
+
+=========================================================*/
+/*-- compiler bug, floating constant overflow w/ansi --*/
+#ifdef _MSC_VER
+#pragma warning(disable:4056)
+#endif
+
+
+
+/* Read Only */
+static long steps[18] =
+{
+   0, 3, 5, 7, 9, 15, 31, 63, 127,
+   255, 511, 1023, 2047, 4095, 8191, 16383, 32767, 65535};
+
+
+/* ABCD_INDEX = lookqt[mode][sr_index][br_index]  */
+/* -1 = invalid  */
+/* Read Only */
+static signed char lookqt[4][3][16] =
+{
+ {{1, -1, -1, -1, 2, -1, 2, 0, 0, 0, 1, 1, 1, 1, 1, -1},	/*  44ks stereo */
+  {0, -1, -1, -1, 2, -1, 2, 0, 0, 0, 0, 0, 0, 0, 0, -1},	/*  48ks */
+  {1, -1, -1, -1, 3, -1, 3, 0, 0, 0, 1, 1, 1, 1, 1, -1}},	/*  32ks */
+ {{1, -1, -1, -1, 2, -1, 2, 0, 0, 0, 1, 1, 1, 1, 1, -1},	/*  44ks joint stereo */
+  {0, -1, -1, -1, 2, -1, 2, 0, 0, 0, 0, 0, 0, 0, 0, -1},	/*  48ks */
+  {1, -1, -1, -1, 3, -1, 3, 0, 0, 0, 1, 1, 1, 1, 1, -1}},	/*  32ks */
+ {{1, -1, -1, -1, 2, -1, 2, 0, 0, 0, 1, 1, 1, 1, 1, -1},	/*  44ks dual chan */
+  {0, -1, -1, -1, 2, -1, 2, 0, 0, 0, 0, 0, 0, 0, 0, -1},	/*  48ks */
+  {1, -1, -1, -1, 3, -1, 3, 0, 0, 0, 1, 1, 1, 1, 1, -1}},	/*  32ks */
+// mono extended beyond legal br index
+//  1,2,2,0,0,0,1,1,1,1,1,1,1,1,1,-1,          /*  44ks single chan */
+//  0,2,2,0,0,0,0,0,0,0,0,0,0,0,0,-1,          /*  48ks */
+//  1,3,3,0,0,0,1,1,1,1,1,1,1,1,1,-1,          /*  32ks */
+// legal mono
+ {{1, 2, 2, 0, 0, 0, 1, 1, 1, 1, 1, -1, -1, -1, -1, -1},	/*  44ks single chan */
+  {0, 2, 2, 0, 0, 0, 0, 0, 0, 0, 0, -1, -1, -1, -1, -1},	/*  48ks */
+  {1, 3, 3, 0, 0, 0, 1, 1, 1, 1, 1, -1, -1, -1, -1, -1}},	/*  32ks */
+};
+/* Read Only */
+static long sr_table[8] =
+{22050L, 24000L, 16000L, 1L,
+ 44100L, 48000L, 32000L, 1L};
+
+/* bit allocation table look up */
+/* table per mpeg spec tables 3b2a/b/c/d  /e is mpeg2 */
+/* look_bat[abcd_index][4][16]  */
+/* Read Only */
+static unsigned char look_bat[5][4][16] =
+{
+/* LOOK_BATA */
+ {{0, 1, 3, 5, 6, 7, 8, 9, 10, 11, 12, 13, 14, 15, 16, 17},
+  {0, 1, 2, 3, 4, 5, 6, 7, 8, 9, 10, 11, 12, 13, 14, 17},
+  {0, 1, 2, 3, 4, 5, 6, 17, 0, 0, 0, 0, 0, 0, 0, 0},
+  {0, 1, 2, 17, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0}},
+/* LOOK_BATB */
+ {{0, 1, 3, 5, 6, 7, 8, 9, 10, 11, 12, 13, 14, 15, 16, 17},
+  {0, 1, 2, 3, 4, 5, 6, 7, 8, 9, 10, 11, 12, 13, 14, 17},
+  {0, 1, 2, 3, 4, 5, 6, 17, 0, 0, 0, 0, 0, 0, 0, 0},
+  {0, 1, 2, 17, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0}},
+/* LOOK_BATC */
+ {{0, 1, 2, 4, 5, 6, 7, 8, 9, 10, 11, 12, 13, 14, 15, 16},
+  {0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0},
+  {0, 1, 2, 4, 5, 6, 7, 8, 0, 0, 0, 0, 0, 0, 0, 0},
+  {0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0}},
+/* LOOK_BATD */
+ {{0, 1, 2, 4, 5, 6, 7, 8, 9, 10, 11, 12, 13, 14, 15, 16},
+  {0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0},
+  {0, 1, 2, 4, 5, 6, 7, 8, 0, 0, 0, 0, 0, 0, 0, 0},
+  {0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0}},
+/* LOOK_BATE */
+ {{0, 1, 2, 3, 4, 5, 6, 7, 8, 9, 10, 11, 12, 13, 14, 15},
+  {0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0},
+  {0, 1, 2, 4, 5, 6, 7, 8, 0, 0, 0, 0, 0, 0, 0, 0},
+  {0, 1, 2, 4, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0}},
+};
+
+/* look_nbat[abcd_index]][4] */
+/* Read Only */
+static unsigned char look_nbat[5][4] =
+{
+  {3, 8, 12, 4},
+  {3, 8, 12, 7},
+  {2, 0, 6, 0},
+  {2, 0, 10, 0},
+  {4, 0, 7, 19},
+};
+
+
+void sbt_mono(MPEG *m, float *sample, short *pcm, int n);
+void sbt_dual(MPEG *m, float *sample, short *pcm, int n);
+void sbt_dual_mono(MPEG *m, float *sample, short *pcm, int n);
+void sbt_dual_left(MPEG *m, float *sample, short *pcm, int n);
+void sbt_dual_right(MPEG *m, float *sample, short *pcm, int n);
+void sbt16_mono(MPEG *m, float *sample, short *pcm, int n);
+void sbt16_dual(MPEG *m, float *sample, short *pcm, int n);
+void sbt16_dual_mono(MPEG *m, float *sample, short *pcm, int n);
+void sbt16_dual_left(MPEG *m, float *sample, short *pcm, int n);
+void sbt16_dual_right(MPEG *m, float *sample, short *pcm, int n);
+void sbt8_mono(MPEG *m, float *sample, short *pcm, int n);
+void sbt8_dual(MPEG *m, float *sample, short *pcm, int n);
+void sbt8_dual_mono(MPEG *m, float *sample, short *pcm, int n);
+void sbt8_dual_left(MPEG *m, float *sample, short *pcm, int n);
+void sbt8_dual_right(MPEG *m, float *sample, short *pcm, int n);
+
+/*--- 8 bit output ---*/
+void sbtB_mono(MPEG *m, float *sample, unsigned char *pcm, int n);
+void sbtB_dual(MPEG *m, float *sample, unsigned char *pcm, int n);
+void sbtB_dual_mono(MPEG *m, float *sample, unsigned char *pcm, int n);
+void sbtB_dual_left(MPEG *m, float *sample, unsigned char *pcm, int n);
+void sbtB_dual_right(MPEG *m, float *sample, unsigned char *pcm, int n);
+void sbtB16_mono(MPEG *m, float *sample, unsigned char *pcm, int n);
+void sbtB16_dual(MPEG *m, float *sample, unsigned char *pcm, int n);
+void sbtB16_dual_mono(MPEG *m, float *sample, unsigned char *pcm, int n);
+void sbtB16_dual_left(MPEG *m, float *sample, unsigned char *pcm, int n);
+void sbtB16_dual_right(MPEG *m, float *sample, unsigned char *pcm, int n);
+void sbtB8_mono(MPEG *m, float *sample, unsigned char *pcm, int n);
+void sbtB8_dual(MPEG *m, float *sample, unsigned char *pcm, int n);
+void sbtB8_dual_mono(MPEG *m, float *sample, unsigned char *pcm, int n);
+void sbtB8_dual_left(MPEG *m, float *sample, unsigned char *pcm, int n);
+void sbtB8_dual_right(MPEG *m, float *sample, unsigned char *pcm, int n);
+
+
+static SBT_FUNCTION_F sbt_table[2][3][5] =
+{
+ {{sbt_mono, sbt_dual, sbt_dual_mono, sbt_dual_left, sbt_dual_right},
+  {sbt16_mono, sbt16_dual, sbt16_dual_mono, sbt16_dual_left, sbt16_dual_right},
+  {sbt8_mono, sbt8_dual, sbt8_dual_mono, sbt8_dual_left, sbt8_dual_right}},
+ {{(SBT_FUNCTION_F) sbtB_mono,
+   (SBT_FUNCTION_F) sbtB_dual,
+   (SBT_FUNCTION_F) sbtB_dual_mono,
+   (SBT_FUNCTION_F) sbtB_dual_left,
+   (SBT_FUNCTION_F) sbtB_dual_right},
+  {(SBT_FUNCTION_F) sbtB16_mono,
+   (SBT_FUNCTION_F) sbtB16_dual,
+   (SBT_FUNCTION_F) sbtB16_dual_mono,
+   (SBT_FUNCTION_F) sbtB16_dual_left,
+   (SBT_FUNCTION_F) sbtB16_dual_right},
+  {(SBT_FUNCTION_F) sbtB8_mono,
+   (SBT_FUNCTION_F) sbtB8_dual,
+   (SBT_FUNCTION_F) sbtB8_dual_mono,
+   (SBT_FUNCTION_F) sbtB8_dual_left,
+   (SBT_FUNCTION_F) sbtB8_dual_right}},
+};
+
+static int out_chans[5] =
+{1, 2, 1, 1, 1};
+
+
+int audio_decode_initL1(MPEG_HEAD * h, int framebytes_arg,
+		   int reduction_code, int transform_code, int convert_code,
+			int freq_limit);
+void sbt_init();
+
+
+IN_OUT L1audio_decode(void *mv, unsigned char *bs, signed short *pcm);
+IN_OUT L2audio_decode(void *mv, unsigned char *bs, signed short *pcm);
+IN_OUT L3audio_decode(void *mv, unsigned char *bs, unsigned char *pcm);
+static AUDIO_DECODE_ROUTINE decode_routine_table[4] =
+{
+   L2audio_decode,
+   (AUDIO_DECODE_ROUTINE)L3audio_decode,
+   L2audio_decode,
+   L1audio_decode,};
+
+extern void cup3_init(MPEG *m);
+
+void mpeg_init(MPEG *m)
+{
+	memset(m, 0, sizeof(MPEG));
+	m->cup.nsb_limit = 6;
+	m->cup.nbat[0] = 3;
+	m->cup.nbat[1] = 8;
+	m->cup.nbat[3] = 12;
+	m->cup.nbat[4] = 7;
+	m->cup.sbt = sbt_mono;
+	m->cup.first_pass = 1;
+	m->cup.first_pass_L1 = 1;
+	m->cup.audio_decode_routine = L2audio_decode;
+	m->cup.cs_factorL1 = m->cup.cs_factor[0];
+	m->cup.nbatL1 = 32;
+	m->cupl.band_limit = 576;
+	m->cupl.band_limit21 = 567;
+	m->cupl.band_limit12 = 576;
+	m->cupl.band_limit_nsb = 32;
+	m->cupl.nsb_limit=32;
+	m->cup.sample = (float *)&m->cupl.sample;
+	m->csbt.first_pass = 1;
+	cup3_init(m);
+}
+
+/*---------------------------------------------------------*/
+static void table_init(MPEG *m)
+{
+   int i, j;
+   int code;
+
+/*--  c_values (dequant) --*/
+   for (i = 1; i < 18; i++)
+      m->cup.look_c_value[i] = 2.0F / steps[i];
+
+/*--  scale factor table, scale by 32768 for 16 pcm output  --*/
+   for (i = 0; i < 64; i++)
+      m->cup.sf_table[i] = (float) (32768.0 * 2.0 * pow(2.0, -i / 3.0));
+
+/*--  grouped 3 level lookup table 5 bit token --*/
+   for (i = 0; i < 32; i++)
+   {
+      code = i;
+      for (j = 0; j < 3; j++)
+      {
+	 m->cup.group3_table[i][j] = (char) ((code % 3) - 1);
+	 code /= 3;
+      }
+   }
+/*--  grouped 5 level lookup table 7 bit token --*/
+   for (i = 0; i < 128; i++)
+   {
+      code = i;
+      for (j = 0; j < 3; j++)
+      {
+	 m->cup.group5_table[i][j] = (char) ((code % 5) - 2);
+	 code /= 5;
+      }
+   }
+/*--  grouped 9 level lookup table 10 bit token --*/
+   for (i = 0; i < 1024; i++)
+   {
+      code = i;
+      for (j = 0; j < 3; j++)
+      {
+	 m->cup.group9_table[i][j] = (short) ((code % 9) - 4);
+	 code /= 9;
+      }
+   }
+
+
+}
+/*---------------------------------------------------------*/
+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 L3audio_decode_init(MPEG *m, MPEG_HEAD * h, int framebytes_arg,
+		   int reduction_code, int transform_code, int convert_code,
+			int freq_limit);
+
+/*---------------------------------------------------------*/
+/* mpeg_head defined in mhead.h  frame bytes is without pad */
+int audio_decode_init(MPEG *m, MPEG_HEAD * h, int framebytes_arg,
+		   int reduction_code, int transform_code, int convert_code,
+		      int freq_limit)
+{
+   int i, j, k;
+   int abcd_index;
+   long samprate;
+   int limit;
+   int bit_code;
+
+   if (m->cup.first_pass)
+   {
+      table_init(m);
+      m->cup.first_pass = 0;
+   }
+
+/* select decoder routine Layer I,II,III */
+   m->cup.audio_decode_routine = decode_routine_table[h->option & 3];
+
+
+   if (h->option == 3)		/* layer I */
+      return L1audio_decode_init(m, h, framebytes_arg,
+		  reduction_code, transform_code, convert_code, freq_limit);
+
+   if (h->option == 1)		/* layer III */
+      return L3audio_decode_init(m, h, framebytes_arg,
+		  reduction_code, transform_code, convert_code, freq_limit);
+
+
+
+   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 != 2)
+      return 0;			/* layer II only */
+   if (h->sr_index == 3)
+      return 0;			/* reserved */
+
+/* compute abcd index for bit allo table selection */
+   if (h->id)			/* mpeg 1 */
+      abcd_index = lookqt[h->mode][h->sr_index][h->br_index];
+   else
+      abcd_index = 4;		/* mpeg 2 */
+
+   if (abcd_index < 0)
+      return 0;			// fail invalid Layer II bit rate index
+
+   for (i = 0; i < 4; i++)
+      for (j = 0; j < 16; j++)
+	 m->cup.bat[i][j] = look_bat[abcd_index][i][j];
+   for (i = 0; i < 4; i++)
+      m->cup.nbat[i] = look_nbat[abcd_index][i];
+   m->cup.max_sb = m->cup.nbat[0] + m->cup.nbat[1] + m->cup.nbat[2] + m->cup.nbat[3];
+/*----- 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 = 1152 >> reduction_code;
+   if (h->mode != 3)
+   {				/* adjust for 2 channel modes */
+      for (i = 0; i < 4; i++)
+	 m->cup.nbat[i] *= 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 < 2304; i++)
+      m->cup.sample[i] = 0.0F;
+
+
+/* init sub-band transform */
+   sbt_init();
+
+   return 1;
+}
+/*---------------------------------------------------------*/
+void audio_decode_info(MPEG *m, DEC_INFO * info)
+{
+   *info = m->cup.decinfo;		/* info return, call after init */
+}
+/*---------------------------------------------------------*/
+void decode_table_init(MPEG *m)
+{
+/* dummy for asm version compatability */
+}
+/*---------------------------------------------------------*/

codecs/mp3/src/cwin.c

@@ -0,0 +1,500 @@
+/*____________________________________________________________________________
+	
+	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.
+	
+	$Id$
+____________________________________________________________________________*/
+
+/****  cwin.c  ***************************************************
+
+include to cwinm.c
+
+MPEG audio decoder, float window routines
+portable C
+
+******************************************************************/
+
+#ifdef ASM_X86
+extern void window_mpg_asm(float *a, int b, short *c);
+extern void window_dual_asm(float *a, int b, short *c);
+extern void window16_asm(float *a, int b, short *c);
+extern void window16_dual_asm(float *a, int b, short *c);
+extern void window8_asm(float *a, int b, short *c);
+extern void window8_dual_asm(float *a, int b, short *c);
+#endif /* ASM_X86 */
+
+/*-------------------------------------------------------------------------*/
+void window(float *vbuf, int vb_ptr, short *pcm)
+{
+#ifdef ASM_X86
+   window_mpg_asm(vbuf, vb_ptr, pcm);
+#else
+   int i, j;
+   int si, bx;
+   float *coef;
+   float sum;
+   long tmp;
+
+   si = vb_ptr + 16;
+   bx = (si + 32) & 511;
+   coef = wincoef;
+
+/*-- first 16 --*/
+   for (i = 0; i < 16; i++)
+   {
+      sum = 0.0F;
+      for (j = 0; j < 8; j++)
+      {
+	 sum += (*coef++) * vbuf[si];
+	 si = (si + 64) & 511;
+	 sum -= (*coef++) * vbuf[bx];
+	 bx = (bx + 64) & 511;
+      }
+      si++;
+      bx--;
+      tmp = (long) sum;
+      if (tmp > 32767)
+	 tmp = 32767;
+      else if (tmp < -32768)
+	 tmp = -32768;
+      *pcm++ = tmp;
+   }
+/*--  special case --*/
+   sum = 0.0F;
+   for (j = 0; j < 8; j++)
+   {
+      sum += (*coef++) * vbuf[bx];
+      bx = (bx + 64) & 511;
+   }
+   tmp = (long) sum;
+   if (tmp > 32767)
+      tmp = 32767;
+   else if (tmp < -32768)
+      tmp = -32768;
+   *pcm++ = tmp;
+/*-- last 15 --*/
+   coef = wincoef + 255;	/* back pass through coefs */
+   for (i = 0; i < 15; i++)
+   {
+      si--;
+      bx++;
+      sum = 0.0F;
+      for (j = 0; j < 8; j++)
+      {
+	 sum += (*coef--) * vbuf[si];
+	 si = (si + 64) & 511;
+	 sum += (*coef--) * vbuf[bx];
+	 bx = (bx + 64) & 511;
+      }
+      tmp = (long) sum;
+      if (tmp > 32767)
+	 tmp = 32767;
+      else if (tmp < -32768)
+	 tmp = -32768;
+      *pcm++ = tmp;
+   }
+#endif
+}
+
+
+
+/*------------------------------------------------------------*/
+#ifndef ASM_X86_OLD
+void window_dual(float *vbuf, int vb_ptr, short *pcm)
+{
+#ifdef ASM_X86
+   window_dual_asm(vbuf, vb_ptr, pcm);
+#else
+   int i, j;			/* dual window interleaves output */
+   int si, bx;
+   float *coef;
+   float sum;
+   long tmp;
+
+   si = vb_ptr + 16;
+   bx = (si + 32) & 511;
+   coef = wincoef;
+
+/*-- first 16 --*/
+   for (i = 0; i < 16; i++)
+   {
+      sum = 0.0F;
+      for (j = 0; j < 8; j++)
+      {
+	 sum += (*coef++) * vbuf[si];
+	 si = (si + 64) & 511;
+	 sum -= (*coef++) * vbuf[bx];
+	 bx = (bx + 64) & 511;
+      }
+      si++;
+      bx--;
+      tmp = (long) sum;
+      if (tmp > 32767)
+	 tmp = 32767;
+      else if (tmp < -32768)
+	 tmp = -32768;
+      *pcm = tmp;
+      pcm += 2;
+   }
+/*--  special case --*/
+   sum = 0.0F;
+   for (j = 0; j < 8; j++)
+   {
+      sum += (*coef++) * vbuf[bx];
+      bx = (bx + 64) & 511;
+   }
+   tmp = (long) sum;
+   if (tmp > 32767)
+      tmp = 32767;
+   else if (tmp < -32768)
+      tmp = -32768;
+   *pcm = tmp;
+   pcm += 2;
+/*-- last 15 --*/
+   coef = wincoef + 255;	/* back pass through coefs */
+   for (i = 0; i < 15; i++)
+   {
+      si--;
+      bx++;
+      sum = 0.0F;
+      for (j = 0; j < 8; j++)
+      {
+	 sum += (*coef--) * vbuf[si];
+	 si = (si + 64) & 511;
+	 sum += (*coef--) * vbuf[bx];
+	 bx = (bx + 64) & 511;
+      }
+      tmp = (long) sum;
+      if (tmp > 32767)
+	 tmp = 32767;
+      else if (tmp < -32768)
+	 tmp = -32768;
+      *pcm = tmp;
+      pcm += 2;
+   }
+#endif
+}
+#endif	/* ndef ASM_X86_OLD */
+/*------------------------------------------------------------*/
+/*------------------- 16 pt window ------------------------------*/
+void window16(float *vbuf, int vb_ptr, short *pcm)
+{
+#ifdef ASM_X86
+   window16_asm(vbuf, vb_ptr, pcm);
+#else
+   int i, j;
+   unsigned char si, bx;
+   float *coef;
+   float sum;
+   long tmp;
+
+   si = vb_ptr + 8;
+   bx = si + 16;
+   coef = wincoef;
+
+/*-- first 8 --*/
+   for (i = 0; i < 8; i++)
+   {
+      sum = 0.0F;
+      for (j = 0; j < 8; j++)
+      {
+	 sum += (*coef++) * vbuf[si];
+	 si += 32;
+	 sum -= (*coef++) * vbuf[bx];
+	 bx += 32;
+      }
+      si++;
+      bx--;
+      coef += 16;
+      tmp = (long) sum;
+      if (tmp > 32767)
+	 tmp = 32767;
+      else if (tmp < -32768)
+	 tmp = -32768;
+      *pcm++ = tmp;
+   }
+/*--  special case --*/
+   sum = 0.0F;
+   for (j = 0; j < 8; j++)
+   {
+      sum += (*coef++) * vbuf[bx];
+      bx += 32;
+   }
+   tmp = (long) sum;
+   if (tmp > 32767)
+      tmp = 32767;
+   else if (tmp < -32768)
+      tmp = -32768;
+   *pcm++ = tmp;
+/*-- last 7 --*/
+   coef = wincoef + 255;	/* back pass through coefs */
+   for (i = 0; i < 7; i++)
+   {
+      coef -= 16;
+      si--;
+      bx++;
+      sum = 0.0F;
+      for (j = 0; j < 8; j++)
+      {
+	 sum += (*coef--) * vbuf[si];
+	 si += 32;
+	 sum += (*coef--) * vbuf[bx];
+	 bx += 32;
+      }
+      tmp = (long) sum;
+      if (tmp > 32767)
+	 tmp = 32767;
+      else if (tmp < -32768)
+	 tmp = -32768;
+      *pcm++ = tmp;
+   }
+#endif
+}
+/*--------------- 16 pt dual window (interleaved output) -----------------*/
+void window16_dual(float *vbuf, int vb_ptr, short *pcm)
+{
+#ifdef ASM_X86
+   window16_dual_asm(vbuf, vb_ptr, pcm);
+#else
+   int i, j;
+   unsigned char si, bx;
+   float *coef;
+   float sum;
+   long tmp;
+
+   si = vb_ptr + 8;
+   bx = si + 16;
+   coef = wincoef;
+
+/*-- first 8 --*/
+   for (i = 0; i < 8; i++)
+   {
+      sum = 0.0F;
+      for (j = 0; j < 8; j++)
+      {
+	 sum += (*coef++) * vbuf[si];
+	 si += 32;
+	 sum -= (*coef++) * vbuf[bx];
+	 bx += 32;
+      }
+      si++;
+      bx--;
+      coef += 16;
+      tmp = (long) sum;
+      if (tmp > 32767)
+	 tmp = 32767;
+      else if (tmp < -32768)
+	 tmp = -32768;
+      *pcm = tmp;
+      pcm += 2;
+   }
+/*--  special case --*/
+   sum = 0.0F;
+   for (j = 0; j < 8; j++)
+   {
+      sum += (*coef++) * vbuf[bx];
+      bx += 32;
+   }
+   tmp = (long) sum;
+   if (tmp > 32767)
+      tmp = 32767;
+   else if (tmp < -32768)
+      tmp = -32768;
+   *pcm = tmp;
+   pcm += 2;
+/*-- last 7 --*/
+   coef = wincoef + 255;	/* back pass through coefs */
+   for (i = 0; i < 7; i++)
+   {
+      coef -= 16;
+      si--;
+      bx++;
+      sum = 0.0F;
+      for (j = 0; j < 8; j++)
+      {
+	 sum += (*coef--) * vbuf[si];
+	 si += 32;
+	 sum += (*coef--) * vbuf[bx];
+	 bx += 32;
+      }
+      tmp = (long) sum;
+      if (tmp > 32767)
+	 tmp = 32767;
+      else if (tmp < -32768)
+	 tmp = -32768;
+      *pcm = tmp;
+      pcm += 2;
+   }
+#endif
+}
+/*------------------- 8 pt window ------------------------------*/
+void window8(float *vbuf, int vb_ptr, short *pcm)
+{
+#ifdef ASM_X86
+   window8_asm(vbuf, vb_ptr, pcm);
+#else
+   int i, j;
+   int si, bx;
+   float *coef;
+   float sum;
+   long tmp;
+
+   si = vb_ptr + 4;
+   bx = (si + 8) & 127;
+   coef = wincoef;
+
+/*-- first 4 --*/
+   for (i = 0; i < 4; i++)
+   {
+      sum = 0.0F;
+      for (j = 0; j < 8; j++)
+      {
+	 sum += (*coef++) * vbuf[si];
+	 si = (si + 16) & 127;
+	 sum -= (*coef++) * vbuf[bx];
+	 bx = (bx + 16) & 127;
+      }
+      si++;
+      bx--;
+      coef += 48;
+      tmp = (long) sum;
+      if (tmp > 32767)
+	 tmp = 32767;
+      else if (tmp < -32768)
+	 tmp = -32768;
+      *pcm++ = tmp;
+   }
+/*--  special case --*/
+   sum = 0.0F;
+   for (j = 0; j < 8; j++)
+   {
+      sum += (*coef++) * vbuf[bx];
+      bx = (bx + 16) & 127;
+   }
+   tmp = (long) sum;
+   if (tmp > 32767)
+      tmp = 32767;
+   else if (tmp < -32768)
+      tmp = -32768;
+   *pcm++ = tmp;
+/*-- last 3 --*/
+   coef = wincoef + 255;	/* back pass through coefs */
+   for (i = 0; i < 3; i++)
+   {
+      coef -= 48;
+      si--;
+      bx++;
+      sum = 0.0F;
+      for (j = 0; j < 8; j++)
+      {
+	 sum += (*coef--) * vbuf[si];
+	 si = (si + 16) & 127;
+	 sum += (*coef--) * vbuf[bx];
+	 bx = (bx + 16) & 127;
+      }
+      tmp = (long) sum;
+      if (tmp > 32767)
+	 tmp = 32767;
+      else if (tmp < -32768)
+	 tmp = -32768;
+      *pcm++ = tmp;
+   }
+#endif
+}
+/*--------------- 8 pt dual window (interleaved output) -----------------*/
+void window8_dual(float *vbuf, int vb_ptr, short *pcm)
+{
+#ifdef ASM_X86
+   window8_dual_asm(vbuf, vb_ptr, pcm);
+#else
+   int i, j;
+   int si, bx;
+   float *coef;
+   float sum;
+   long tmp;
+
+   si = vb_ptr + 4;
+   bx = (si + 8) & 127;
+   coef = wincoef;
+
+/*-- first 4 --*/
+   for (i = 0; i < 4; i++)
+   {
+      sum = 0.0F;
+      for (j = 0; j < 8; j++)
+      {
+	 sum += (*coef++) * vbuf[si];
+	 si = (si + 16) & 127;
+	 sum -= (*coef++) * vbuf[bx];
+	 bx = (bx + 16) & 127;
+      }
+      si++;
+      bx--;
+      coef += 48;
+      tmp = (long) sum;
+      if (tmp > 32767)
+	 tmp = 32767;
+      else if (tmp < -32768)
+	 tmp = -32768;
+      *pcm = tmp;
+      pcm += 2;
+   }
+/*--  special case --*/
+   sum = 0.0F;
+   for (j = 0; j < 8; j++)
+   {
+      sum += (*coef++) * vbuf[bx];
+      bx = (bx + 16) & 127;
+   }
+   tmp = (long) sum;
+   if (tmp > 32767)
+      tmp = 32767;
+   else if (tmp < -32768)
+      tmp = -32768;
+   *pcm = tmp;
+   pcm += 2;
+/*-- last 3 --*/
+   coef = wincoef + 255;	/* back pass through coefs */
+   for (i = 0; i < 3; i++)
+   {
+      coef -= 48;
+      si--;
+      bx++;
+      sum = 0.0F;
+      for (j = 0; j < 8; j++)
+      {
+	 sum += (*coef--) * vbuf[si];
+	 si = (si + 16) & 127;
+	 sum += (*coef--) * vbuf[bx];
+	 bx = (bx + 16) & 127;
+      }
+      tmp = (long) sum;
+      if (tmp > 32767)
+	 tmp = 32767;
+      else if (tmp < -32768)
+	 tmp = -32768;
+      *pcm = tmp;
+      pcm += 2;
+   }
+#endif
+}
+/*------------------------------------------------------------*/

codecs/mp3/src/cwinb.c

@@ -0,0 +1,496 @@
+/*____________________________________________________________________________
+	
+	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.
+	
+	$Id$
+____________________________________________________________________________*/
+
+/****  cwin.c  ***************************************************
+
+include to cwinm.c
+
+MPEG audio decoder, float window routines - 8 bit output
+portable C
+
+******************************************************************/
+/*-------------------------------------------------------------------------*/
+
+
+#ifdef ASM_X86
+extern void windowB_asm(float *a, int b, unsigned char *c);
+extern void windowB_dual_asm(float *a, int b, unsigned char *c);
+extern void windowB16_asm(float *a, int b, unsigned char *c);
+extern void windowB16_dual_asm(float *a, int b, unsigned char *c);
+extern void windowB8_asm(float *a, int b, unsigned char *c);
+extern void windowB8_dual_asm(float *a, int b, unsigned char *c);
+#endif /* ASM_X86 */
+
+void windowB(float *vbuf, int vb_ptr, unsigned char *pcm)
+{
+#ifdef ASM_X86
+   windowB_asm(vbuf, vb_ptr, pcm);
+#else
+   int i, j;
+   int si, bx;
+   float *coef;
+   float sum;
+   long tmp;
+
+   si = vb_ptr + 16;
+   bx = (si + 32) & 511;
+   coef = wincoef;
+
+/*-- first 16 --*/
+   for (i = 0; i < 16; i++)
+   {
+      sum = 0.0F;
+      for (j = 0; j < 8; j++)
+      {
+	 sum += (*coef++) * vbuf[si];
+	 si = (si + 64) & 511;
+	 sum -= (*coef++) * vbuf[bx];
+	 bx = (bx + 64) & 511;
+      }
+      si++;
+      bx--;
+      tmp = (long) sum;
+      if (tmp > 32767)
+	 tmp = 32767;
+      else if (tmp < -32768)
+	 tmp = -32768;
+      *pcm++ = ((unsigned char) (tmp >> 8)) ^ 0x80;
+   }
+/*--  special case --*/
+   sum = 0.0F;
+   for (j = 0; j < 8; j++)
+   {
+      sum += (*coef++) * vbuf[bx];
+      bx = (bx + 64) & 511;
+   }
+   tmp = (long) sum;
+   if (tmp > 32767)
+      tmp = 32767;
+   else if (tmp < -32768)
+      tmp = -32768;
+   *pcm++ = ((unsigned char) (tmp >> 8)) ^ 0x80;
+/*-- last 15 --*/
+   coef = wincoef + 255;	/* back pass through coefs */
+   for (i = 0; i < 15; i++)
+   {
+      si--;
+      bx++;
+      sum = 0.0F;
+      for (j = 0; j < 8; j++)
+      {
+	 sum += (*coef--) * vbuf[si];
+	 si = (si + 64) & 511;
+	 sum += (*coef--) * vbuf[bx];
+	 bx = (bx + 64) & 511;
+      }
+      tmp = (long) sum;
+      if (tmp > 32767)
+	 tmp = 32767;
+      else if (tmp < -32768)
+	 tmp = -32768;
+      *pcm++ = ((unsigned char) (tmp >> 8)) ^ 0x80;
+   }
+#endif
+}
+/*------------------------------------------------------------*/
+void windowB_dual(float *vbuf, int vb_ptr, unsigned char *pcm)
+{
+#ifdef ASM_X86
+   windowB_dual_asm(vbuf, vb_ptr, pcm);
+#else
+   int i, j;			/* dual window interleaves output */
+   int si, bx;
+   float *coef;
+   float sum;
+   long tmp;
+
+   si = vb_ptr + 16;
+   bx = (si + 32) & 511;
+   coef = wincoef;
+
+/*-- first 16 --*/
+   for (i = 0; i < 16; i++)
+   {
+      sum = 0.0F;
+      for (j = 0; j < 8; j++)
+      {
+	 sum += (*coef++) * vbuf[si];
+	 si = (si + 64) & 511;
+	 sum -= (*coef++) * vbuf[bx];
+	 bx = (bx + 64) & 511;
+      }
+      si++;
+      bx--;
+      tmp = (long) sum;
+      if (tmp > 32767)
+	 tmp = 32767;
+      else if (tmp < -32768)
+	 tmp = -32768;
+      *pcm = ((unsigned char) (tmp >> 8)) ^ 0x80;
+      pcm += 2;
+   }
+/*--  special case --*/
+   sum = 0.0F;
+   for (j = 0; j < 8; j++)
+   {
+      sum += (*coef++) * vbuf[bx];
+      bx = (bx + 64) & 511;
+   }
+   tmp = (long) sum;
+   if (tmp > 32767)
+      tmp = 32767;
+   else if (tmp < -32768)
+      tmp = -32768;
+   *pcm = ((unsigned char) (tmp >> 8)) ^ 0x80;
+   pcm += 2;
+/*-- last 15 --*/
+   coef = wincoef + 255;	/* back pass through coefs */
+   for (i = 0; i < 15; i++)
+   {
+      si--;
+      bx++;
+      sum = 0.0F;
+      for (j = 0; j < 8; j++)
+      {
+	 sum += (*coef--) * vbuf[si];
+	 si = (si + 64) & 511;
+	 sum += (*coef--) * vbuf[bx];
+	 bx = (bx + 64) & 511;
+      }
+      tmp = (long) sum;
+      if (tmp > 32767)
+	 tmp = 32767;
+      else if (tmp < -32768)
+	 tmp = -32768;
+      *pcm = ((unsigned char) (tmp >> 8)) ^ 0x80;
+      pcm += 2;
+   }
+#endif
+}
+/*------------------------------------------------------------*/
+/*------------------- 16 pt window ------------------------------*/
+void windowB16(float *vbuf, int vb_ptr, unsigned char *pcm)
+{
+#ifdef ASM_X86
+   windowB16_asm(vbuf, vb_ptr, pcm);
+#else
+   int i, j;
+   unsigned char si, bx;
+   float *coef;
+   float sum;
+   long tmp;
+
+   si = vb_ptr + 8;
+   bx = si + 16;
+   coef = wincoef;
+
+/*-- first 8 --*/
+   for (i = 0; i < 8; i++)
+   {
+      sum = 0.0F;
+      for (j = 0; j < 8; j++)
+      {
+	 sum += (*coef++) * vbuf[si];
+	 si += 32;
+	 sum -= (*coef++) * vbuf[bx];
+	 bx += 32;
+      }
+      si++;
+      bx--;
+      coef += 16;
+      tmp = (long) sum;
+      if (tmp > 32767)
+	 tmp = 32767;
+      else if (tmp < -32768)
+	 tmp = -32768;
+      *pcm++ = ((unsigned char) (tmp >> 8)) ^ 0x80;
+   }
+/*--  special case --*/
+   sum = 0.0F;
+   for (j = 0; j < 8; j++)
+   {
+      sum += (*coef++) * vbuf[bx];
+      bx += 32;
+   }
+   tmp = (long) sum;
+   if (tmp > 32767)
+      tmp = 32767;
+   else if (tmp < -32768)
+      tmp = -32768;
+   *pcm++ = ((unsigned char) (tmp >> 8)) ^ 0x80;
+/*-- last 7 --*/
+   coef = wincoef + 255;	/* back pass through coefs */
+   for (i = 0; i < 7; i++)
+   {
+      coef -= 16;
+      si--;
+      bx++;
+      sum = 0.0F;
+      for (j = 0; j < 8; j++)
+      {
+	 sum += (*coef--) * vbuf[si];
+	 si += 32;
+	 sum += (*coef--) * vbuf[bx];
+	 bx += 32;
+      }
+      tmp = (long) sum;
+      if (tmp > 32767)
+	 tmp = 32767;
+      else if (tmp < -32768)
+	 tmp = -32768;
+      *pcm++ = ((unsigned char) (tmp >> 8)) ^ 0x80;
+   }
+#endif
+}
+/*--------------- 16 pt dual window (interleaved output) -----------------*/
+void windowB16_dual(float *vbuf, int vb_ptr, unsigned char *pcm)
+{
+#ifdef ASM_X86
+   windowB16_dual_asm(vbuf, vb_ptr, pcm);
+#else
+   int i, j;
+   unsigned char si, bx;
+   float *coef;
+   float sum;
+   long tmp;
+
+   si = vb_ptr + 8;
+   bx = si + 16;
+   coef = wincoef;
+
+/*-- first 8 --*/
+   for (i = 0; i < 8; i++)
+   {
+      sum = 0.0F;
+      for (j = 0; j < 8; j++)
+      {
+	 sum += (*coef++) * vbuf[si];
+	 si += 32;
+	 sum -= (*coef++) * vbuf[bx];
+	 bx += 32;
+      }
+      si++;
+      bx--;
+      coef += 16;
+      tmp = (long) sum;
+      if (tmp > 32767)
+	 tmp = 32767;
+      else if (tmp < -32768)
+	 tmp = -32768;
+      *pcm = ((unsigned char) (tmp >> 8)) ^ 0x80;
+      pcm += 2;
+   }
+/*--  special case --*/
+   sum = 0.0F;
+   for (j = 0; j < 8; j++)
+   {
+      sum += (*coef++) * vbuf[bx];
+      bx += 32;
+   }
+   tmp = (long) sum;
+   if (tmp > 32767)
+      tmp = 32767;
+   else if (tmp < -32768)
+      tmp = -32768;
+   *pcm = ((unsigned char) (tmp >> 8)) ^ 0x80;
+   pcm += 2;
+/*-- last 7 --*/
+   coef = wincoef + 255;	/* back pass through coefs */
+   for (i = 0; i < 7; i++)
+   {
+      coef -= 16;
+      si--;
+      bx++;
+      sum = 0.0F;
+      for (j = 0; j < 8; j++)
+      {
+	 sum += (*coef--) * vbuf[si];
+	 si += 32;
+	 sum += (*coef--) * vbuf[bx];
+	 bx += 32;
+      }
+      tmp = (long) sum;
+      if (tmp > 32767)
+	 tmp = 32767;
+      else if (tmp < -32768)
+	 tmp = -32768;
+      *pcm = ((unsigned char) (tmp >> 8)) ^ 0x80;
+      pcm += 2;
+   }
+#endif
+}
+/*------------------- 8 pt window ------------------------------*/
+void windowB8(float *vbuf, int vb_ptr, unsigned char *pcm)
+{
+#ifdef ASM_X86
+   windowB8_asm(vbuf, vb_ptr, pcm);
+#else
+   int i, j;
+   int si, bx;
+   float *coef;
+   float sum;
+   long tmp;
+
+   si = vb_ptr + 4;
+   bx = (si + 8) & 127;
+   coef = wincoef;
+
+/*-- first 4 --*/
+   for (i = 0; i < 4; i++)
+   {
+      sum = 0.0F;
+      for (j = 0; j < 8; j++)
+      {
+	 sum += (*coef++) * vbuf[si];
+	 si = (si + 16) & 127;
+	 sum -= (*coef++) * vbuf[bx];
+	 bx = (bx + 16) & 127;
+      }
+      si++;
+      bx--;
+      coef += 48;
+      tmp = (long) sum;
+      if (tmp > 32767)
+	 tmp = 32767;
+      else if (tmp < -32768)
+	 tmp = -32768;
+      *pcm++ = ((unsigned char) (tmp >> 8)) ^ 0x80;
+   }
+/*--  special case --*/
+   sum = 0.0F;
+   for (j = 0; j < 8; j++)
+   {
+      sum += (*coef++) * vbuf[bx];
+      bx = (bx + 16) & 127;
+   }
+   tmp = (long) sum;
+   if (tmp > 32767)
+      tmp = 32767;
+   else if (tmp < -32768)
+      tmp = -32768;
+   *pcm++ = ((unsigned char) (tmp >> 8)) ^ 0x80;
+/*-- last 3 --*/
+   coef = wincoef + 255;	/* back pass through coefs */
+   for (i = 0; i < 3; i++)
+   {
+      coef -= 48;
+      si--;
+      bx++;
+      sum = 0.0F;
+      for (j = 0; j < 8; j++)
+      {
+	 sum += (*coef--) * vbuf[si];
+	 si = (si + 16) & 127;
+	 sum += (*coef--) * vbuf[bx];
+	 bx = (bx + 16) & 127;
+      }
+      tmp = (long) sum;
+      if (tmp > 32767)
+	 tmp = 32767;
+      else if (tmp < -32768)
+	 tmp = -32768;
+      *pcm++ = ((unsigned char) (tmp >> 8)) ^ 0x80;
+   }
+#endif
+}
+/*--------------- 8 pt dual window (interleaved output) -----------------*/
+void windowB8_dual(float *vbuf, int vb_ptr, unsigned char *pcm)
+{
+#ifdef ASM_X86
+   windowB8_dual_asm(vbuf, vb_ptr, pcm);
+#else
+   int i, j;
+   int si, bx;
+   float *coef;
+   float sum;
+   long tmp;
+
+   si = vb_ptr + 4;
+   bx = (si + 8) & 127;
+   coef = wincoef;
+
+/*-- first 4 --*/
+   for (i = 0; i < 4; i++)
+   {
+      sum = 0.0F;
+      for (j = 0; j < 8; j++)
+      {
+	 sum += (*coef++) * vbuf[si];
+	 si = (si + 16) & 127;
+	 sum -= (*coef++) * vbuf[bx];
+	 bx = (bx + 16) & 127;
+      }
+      si++;
+      bx--;
+      coef += 48;
+      tmp = (long) sum;
+      if (tmp > 32767)
+	 tmp = 32767;
+      else if (tmp < -32768)
+	 tmp = -32768;
+      *pcm = ((unsigned char) (tmp >> 8)) ^ 0x80;
+      pcm += 2;
+   }
+/*--  special case --*/
+   sum = 0.0F;
+   for (j = 0; j < 8; j++)
+   {
+      sum += (*coef++) * vbuf[bx];
+      bx = (bx + 16) & 127;
+   }
+   tmp = (long) sum;
+   if (tmp > 32767)
+      tmp = 32767;
+   else if (tmp < -32768)
+      tmp = -32768;
+   *pcm = ((unsigned char) (tmp >> 8)) ^ 0x80;
+   pcm += 2;
+/*-- last 3 --*/
+   coef = wincoef + 255;	/* back pass through coefs */
+   for (i = 0; i < 3; i++)
+   {
+      coef -= 48;
+      si--;
+      bx++;
+      sum = 0.0F;
+      for (j = 0; j < 8; j++)
+      {
+	 sum += (*coef--) * vbuf[si];
+	 si = (si + 16) & 127;
+	 sum += (*coef--) * vbuf[bx];
+	 bx = (bx + 16) & 127;
+      }
+      tmp = (long) sum;
+      if (tmp > 32767)
+	 tmp = 32767;
+      else if (tmp < -32768)
+	 tmp = -32768;
+      *pcm = ((unsigned char) (tmp >> 8)) ^ 0x80;
+      pcm += 2;
+   }
+#endif
+}
+/*------------------------------------------------------------*/

codecs/mp3/src/cwinm.c

@@ -0,0 +1,59 @@
+/*____________________________________________________________________________
+	
+	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.
+	
+	$Id$
+____________________________________________________________________________*/
+
+/****  cwinm.c  ***************************************************
+
+MPEG audio decoder, window master routine
+portable C
+
+
+******************************************************************/
+
+#include <stdlib.h>
+#include <stdio.h>
+#include <float.h>
+#include <math.h>
+
+
+/* disable precision loss warning on type conversion */
+#ifdef _MSC_VER
+#pragma warning(disable:4244 4056)
+#endif
+
+float wincoef[264] =
+{				/* window coefs */
+#include "tableawd.h"
+};
+
+/*--------------------------------------------------------*/
+#ifdef QUICK_FLOAT
+#include "cwinq.c"
+#include "cwinbq.c"
+#else
+#include "cwin.c"
+#include "cwinb.c"
+#endif
+/*--------------------------------------------------------*/

codecs/mp3/src/dec8.c

@@ -0,0 +1,338 @@
+/*____________________________________________________________________________
+	
+	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.
+	
+	$Id$
+____________________________________________________________________________*/
+
+/****  dec8.c  ***************************************************
+
+
+ANSI C
+MPEG audio decoder Layer II only mpeg1 and mpeg2
+output sample type and sample rate conversion
+  decode mpeg to 8000Ks mono
+  output 16 bit linear, 8 bit linear, or u-law
+
+
+mod 6/29/95  bugfix in u-law table
+
+mod 11/15/95 for Layer I
+
+mod 1/7/97 minor mods for warnings  
+
+******************************************************************/
+/*****************************************************************
+
+       MPEG audio software decoder portable ANSI c.
+       Decodes all Layer II to 8000Ks mono pcm.
+       Output selectable: 16 bit linear, 8 bit linear, u-law.
+
+-------------------------------------
+int audio_decode8_init(MPEG_HEAD *h, int framebytes_arg,
+         int reduction_code, int transform_code, int convert_code,
+         int freq_limit)
+
+initilize decoder:
+       return 0 = fail, not 0 = success
+
+MPEG_HEAD *h    input, mpeg header info (returned by call to head_info)
+framebytes      input, mpeg frame size (returned by call to head_info)
+reduction_code  input, ignored
+transform_code  input, ignored
+convert_code    input, set convert_code = 4*bit_code + chan_code
+                     bit_code:   1 = 16 bit linear pcm
+                                 2 =  8 bit (unsigned) linear pcm
+                                 3 = u-law (8 bits unsigned)
+                     chan_code:  0 = convert two chan to mono
+                                 1 = convert two chan to mono
+                                 2 = convert two chan to left chan
+                                 3 = convert two chan to right chan
+freq_limit      input, ignored
+
+
+---------------------------------
+void audio_decode8_info( DEC_INFO *info)
+
+information return:
+          Call after audio_decode8_init.  See mhead.h for
+          information returned in DEC_INFO structure.
+
+
+---------------------------------
+IN_OUT audio_decode8(unsigned char *bs, void *pcmbuf)
+
+decode one mpeg audio frame:
+bs        input, mpeg bitstream, must start with
+          sync word.  Caution: may read up to 3 bytes
+          beyond end of frame.
+pcmbuf    output, pcm samples.
+
+IN_OUT structure returns:
+          Number bytes conceptually removed from mpeg bitstream.
+          Returns 0 if sync loss.
+          Number bytes of pcm output.  This may vary from frame
+          to frame.
+
+*****************************************************************/
+
+#include <stdlib.h>
+#include <stdio.h>
+#include <float.h>
+#include <math.h>
+#include "L3.h"
+#include "mhead.h"		/* mpeg header structure */
+
+
+
+
+
+/*------------------------------------------*/
+static int output_code;
+static int convert(void *mv, unsigned char *pcm);
+static int convert_8bit(void *mv, unsigned char *pcm);
+static int convert_u(void *mv, unsigned char *pcm);
+static CVT_FUNCTION_8 cvt_table[3] =
+{
+   convert,
+   convert_8bit,
+   convert_u,
+};
+
+void mpeg8_init(MPEG8 *m)
+{
+	memset(&m->dec, 0, sizeof(m->dec));
+	m->dec.ncnt = 8 * 288;
+	m->dec.ncnt1 = 8 * 287;
+	m->dec.nlast = 287;
+	m->dec.ndeci = 11;
+	m->dec.kdeci = 8 * 288;
+	m->dec.first_pass = 1;
+}
+
+/*====================================================================*/
+IN_OUT audio_decode8(MPEG8 *m, unsigned char *bs, signed short *pcmbuf)
+{
+   IN_OUT x;
+
+   x = audio_decode(&m->cupper, bs, m->dec.pcm);
+   if (x.in_bytes <= 0)
+      return x;
+   x.out_bytes = m->dec.convert_routine(m, (void *) pcmbuf);
+
+   return x;
+}
+/*--------------8Ks 16 bit pcm --------------------------------*/
+static int convert(void *mv, unsigned char y0[])
+{
+	MPEG8 *m = mv;
+   int i, k;
+   long alpha;
+   short *y;
+
+   y = (short *) y0;
+   k = 0;
+   if (m->dec.kdeci < m->dec.ncnt)
+   {
+      alpha = m->dec.kdeci & 7;
+      y[k++] = (short) (m->dec.xsave + ((alpha * (m->dec.pcm[0] - m->dec.xsave)) >> 3));
+      m->dec.kdeci += m->dec.ndeci;
+   }
+   m->dec.kdeci -= m->dec.ncnt;
+   for (; m->dec.kdeci < m->dec.ncnt1; m->dec.kdeci += m->dec.ndeci)
+   {
+      i = m->dec.kdeci >> 3;
+      alpha = m->dec.kdeci & 7;
+      y[k++] = (short) (m->dec.pcm[i] + ((alpha * (m->dec.pcm[i + 1] - m->dec.pcm[i])) >> 3));
+   }
+   m->dec.xsave = m->dec.pcm[m->dec.nlast];
+
+/* printf("\n k out = %4d", k);   */
+
+   return sizeof(short) * k;
+}
+/*----------------8Ks 8 bit unsigned pcm ---------------------------*/
+static int convert_8bit(void *mv, unsigned char y[])
+{
+   MPEG8 *m = mv;
+   int i, k;
+   long alpha;
+
+   k = 0;
+   if (m->dec.kdeci < m->dec.ncnt)
+   {
+      alpha = m->dec.kdeci & 7;
+      y[k++] = (unsigned char) (((m->dec.xsave + ((alpha * (m->dec.pcm[0] - m->dec.xsave)) >> 3)) >> 8) + 128);
+      m->dec.kdeci += m->dec.ndeci;
+   }
+   m->dec.kdeci -= m->dec.ncnt;
+   for (; m->dec.kdeci < m->dec.ncnt1; m->dec.kdeci += m->dec.ndeci)
+   {
+      i = m->dec.kdeci >> 3;
+      alpha = m->dec.kdeci & 7;
+      y[k++] = (unsigned char) (((m->dec.pcm[i] + ((alpha * (m->dec.pcm[i + 1] - m->dec.pcm[i])) >> 3)) >> 8) + 128);
+   }
+   m->dec.xsave = m->dec.pcm[m->dec.nlast];
+
+/* printf("\n k out = %4d", k);   */
+
+   return k;
+}
+/*--------------8Ks u-law --------------------------------*/
+static int convert_u(void *mv, unsigned char y[])
+{
+   MPEG8 *m = mv;
+   int i, k;
+   long alpha;
+   unsigned char *look;
+
+   look = m->dec.look_u + 4096;
+
+   k = 0;
+   if (m->dec.kdeci < m->dec.ncnt)
+   {
+      alpha = m->dec.kdeci & 7;
+      y[k++] = look[(m->dec.xsave + ((alpha * (m->dec.pcm[0] - m->dec.xsave)) >> 3)) >> 3];
+      m->dec.kdeci += m->dec.ndeci;
+   }
+   m->dec.kdeci -= m->dec.ncnt;
+   for (; m->dec.kdeci < m->dec.ncnt1; m->dec.kdeci += m->dec.ndeci)
+   {
+      i = m->dec.kdeci >> 3;
+      alpha = m->dec.kdeci & 7;
+      y[k++] = look[(m->dec.pcm[i] + ((alpha * (m->dec.pcm[i + 1] - m->dec.pcm[i])) >> 3)) >> 3];
+   }
+   m->dec.xsave = m->dec.pcm[m->dec.nlast];
+
+/* printf("\n k out = %4d", k);   */
+
+   return k;
+}
+/*--------------------------------------------------------------------*/
+static int ucomp3(int x)	/* re analog devices CCITT G.711 */
+{
+   int s, p, y, t, u, u0, sign;
+
+   sign = 0;
+   if (x < 0)
+   {
+      x = -x;
+      sign = 0x0080;
+   }
+   if (x > 8031)
+      x = 8031;
+   x += 33;
+   t = x;
+   for (s = 0; s < 15; s++)
+   {
+      if (t & 0x4000)
+	 break;
+      t <<= 1;
+   }
+   y = x << s;
+   p = (y >> 10) & 0x0f;	/* position */
+   s = 9 - s;			/* segment */
+   u0 = (((s << 4) | p) & 0x7f) | sign;
+   u = u0 ^ 0xff;
+
+   return u;
+}
+/*------------------------------------------------------------------*/
+static void table_init(MPEG8 *m)
+{
+   int i;
+
+   for (i = -4096; i < 4096; i++)
+      m->dec.look_u[4096 + i] = (unsigned char) (ucomp3(2 * i));
+
+}
+/*-------------------------------------------------------------------*/
+int audio_decode8_init(MPEG8 *m, MPEG_HEAD * h, int framebytes_arg,
+		   int reduction_code, int transform_code, int convert_code,
+		       int freq_limit)
+{
+   int istat;
+   int outvals;
+   static int sr_table[2][4] =
+   {{22, 24, 16, 0}, {44, 48, 32, 0}};
+
+   if (m->dec.first_pass)
+   {
+      table_init(m);
+      m->dec.first_pass = 0;
+   }
+
+   if ((h->sync & 1) == 0)
+      return 0;			// fail mpeg 2.5
+
+   output_code = convert_code >> 2;
+   if (output_code < 1)
+      output_code = 1;		/* 1= 16bit 2 = 8bit 3 = u */
+   if (output_code > 3)
+      output_code = 3;		/* 1= 16bit 2 = 8bit 3 = u */
+
+   convert_code = convert_code & 3;
+   if (convert_code <= 0)
+      convert_code = 1;		/* always cvt to mono */
+
+   reduction_code = 1;
+   if (h->id)
+      reduction_code = 2;
+
+/* select convert routine */
+   m->dec.convert_routine = cvt_table[output_code - 1];
+
+/* init decimation/convert routine */
+/*-- MPEG-2 layer III --*/
+   if ((h->option == 1) && h->id == 0)
+      outvals = 576 >> reduction_code;
+   else if (h->option == 3)
+      outvals = 384 >> reduction_code;
+/*-- layer I --*/
+   else
+      outvals = 1152 >> reduction_code;
+   m->dec.ncnt = 8 * outvals;
+   m->dec.ncnt1 = 8 * (outvals - 1);
+   m->dec.nlast = outvals - 1;
+   m->dec.ndeci = sr_table[h->id][h->sr_index] >> reduction_code;
+   m->dec.kdeci = 8 * outvals;
+/* printf("\n outvals %d", outvals);  */
+
+   freq_limit = 3200;
+   istat = audio_decode_init(&m->cupper, h, framebytes_arg,
+			     reduction_code, transform_code, convert_code,
+			     freq_limit);
+
+
+   return istat;
+}
+/*-----------------------------------------------------------------*/
+void audio_decode8_info(MPEG8 *m, DEC_INFO * info)
+{
+
+   audio_decode_info(&m->cupper, info);
+   info->samprate = 8000;
+   if (output_code != 1)
+      info->bits = 8;
+   if (output_code == 3)
+      info->type = 10;
+}