Version 0.1.0 from FTP

git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@87 65c4cc65-6c06-0410-ace0-fbb531ad65f3

BUGS

+* EVERYTHING MARKED WITH "XXX" IN THE SOURCE REPRESENTS A BUG!  Sometimes
+  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.

codecs/Makefile

+#
+# Asterisk -- A telephony toolkit for Linux.
+# 
+# Makefile for PBX frontends (dynamically loaded)
+#
+# Copyright (C) 1999, Adtran Inc. and Linux Support Services, LLC
+#
+# Mark Spencer <markster@linux-support.net>
+#
+# This program is free software, distributed under the terms of
+# the GNU General Public License
+#
+
+#
+# Uncomment if you have g723.1 code (with the same API as the Annex-A code
+# and have placed it in the g723.1 directory and/or the Annex-B code in 
+# g723.1b)
+#
+#MODG723=codec_g723_1.so codec_g723_1b.so
+MODG723=$(shell [ -f g723.1/coder.c ] && echo "codec_g723_1.so")
+MODG723+=$(shell [ -f g723.1b/coder2.c ] && echo "codec_g723_1b.so")
+
+CFLAGS+=
+
+LIBG723=g723.1/libg723.a
+LIBG723B=g723.1b/libg723b.a
+LIBGSM=gsm/lib/libgsm.a
+LIBMP3=mp3/libmp3.a
+
+CODECS+=$(MODG723) codec_gsm.so #codec_mp3_d.so
+
+all: $(CODECS)
+
+clean:
+	rm -f *.so *.o
+	make -C g723.1 clean
+	make -C g723.1b clean
+	make -C gsm clean
+	make -C mp3 clean
+
+$(LIBG723):
+	make -C g723.1 all
+
+$(LIBGSM):
+	make -C gsm lib/libgsm.a
+
+$(LIBG723B):
+	make -C g723.1b all
+
+$(LIBMP3):
+	make -C mp3 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 $<
+
+codec_g723_1b.so : codec_g723_1b.o $(LIBG723B)
+	$(CC) -shared -Xlinker -x -o $@ $< $(LIBG723B) -lm
+
+codec_gsm.so: codec_gsm.o $(LIBGSM)
+	$(CC) -shared -Xlinker -x -o $@ $< $(LIBGSM)
+
+codec_mp3_d.so: codec_mp3_d.o $(LIBMP3)
+	$(CC) -shared -Xlinker -x -o $@ $< $(LIBMP3)
+
+%.so : %.o
+	$(CC) -shared -Xlinker -x -o $@ $<
+
+install: all
+	for x in $(CODECS); do $(INSTALL) -m 755 $$x $(MODULES_DIR) ; done

codecs/codec_g723_1.c

+/*
+ * Asterisk -- A telephony toolkit for Linux.
+ *
+ * Translate between signed linear and G.723.1
+ *
+ * The G.723.1 code is not included in the Asterisk distribution because
+ * 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
+ *
+ * Mark Spencer <markster@linux-support.net>
+ *
+ * This program is free software, distributed under the terms of
+ * the GNU General Public License
+ */
+
+#define TYPE_SILENCE	 0x2
+#define TYPE_HIGH	 0x0
+#define TYPE_LOW	 0x1
+#define TYPE_MASK	 0x3
+
+#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>
+
+#ifdef ANNEX_B
+#include "g723.1b/typedef2.h"
+#include "g723.1b/cst2.h"
+#include "g723.1b/coder2.h"
+#include "g723.1b/decod2.h"
+#include "g723.1b/deccng2.h"
+#include "g723.1b/codcng2.h"
+#include "g723.1b/vad2.h"
+#else
+#include "g723.1/typedef.h"
+#include "g723.1/cst_lbc.h"
+#include "g723.1/coder.h"
+#include "g723.1/decod.h"
+#include "g723.1/dec_cng.h"
+#include "g723.1/cod_cng.h"
+#include "g723.1/vad.h"
+#endif
+
+/* Sample frame data */
+#include "slin_g723_ex.h"
+#include "g723_slin_ex.h"
+
+static pthread_mutex_t localuser_lock = PTHREAD_MUTEX_INITIALIZER;
+static int localusecnt=0;
+
+#ifdef ANNEX_B
+static char *tdesc = "Annex B (floating point) G.723.1/PCM16 Codec Translator";
+#else
+static char *tdesc = "Annex A (fixed point) G.723.1/PCM16 Codec Translator";
+#endif
+
+/* Globals */
+Flag UsePf = True;
+Flag UseHp = True;
+Flag UseVx = True;
+
+enum Crate WrkRate = Rate63;
+
+struct g723_encoder_pvt {
+	struct cod_state cod;
+	struct ast_frame f;
+	/* Space to build offset */
+	char offset[AST_FRIENDLY_OFFSET];
+	/* Buffer for our outgoing frame */
+	char outbuf[24];
+	/* Enough to store a full second */
+	short buf[8000];
+	int tail;
+};
+
+struct g723_decoder_pvt {
+	struct dec_state dec;
+	struct ast_frame f;
+	/* Space to build offset */
+	char offset[AST_FRIENDLY_OFFSET];
+	/* Enough to store a full second */
+	short buf[8000];
+	int tail;
+};
+
+static struct ast_translator_pvt *g723tolin_new()
+{
+	struct g723_decoder_pvt *tmp;
+	tmp = malloc(sizeof(struct g723_decoder_pvt));
+	if (tmp) {
+		Init_Decod(&tmp->dec);
+	    Init_Dec_Cng(&tmp->dec);
+		tmp->tail = 0;
+	}
+	return (struct ast_translator_pvt *)tmp;
+}
+
+static struct ast_frame *lintog723_sample()
+{
+	static struct ast_frame f;
+	f.frametype = AST_FRAME_VOICE;
+	f.subclass = AST_FORMAT_SLINEAR;
+	f.datalen = sizeof(slin_g723_ex);
+	/* Assume 8000 Hz */
+	f.timelen = sizeof(slin_g723_ex)/16;
+	f.mallocd = 0;
+	f.offset = 0;
+	f.src = __PRETTY_FUNCTION__;
+	f.data = slin_g723_ex;
+	return &f;
+}
+
+static struct ast_frame *g723tolin_sample()
+{
+	static struct ast_frame f;
+	f.frametype = AST_FRAME_VOICE;
+	f.subclass = AST_FORMAT_G723_1;
+	f.datalen = sizeof(g723_slin_ex);
+	/* All frames are 30 ms long */
+	f.timelen = 30;
+	f.mallocd = 0;
+	f.offset = 0;
+	f.src = __PRETTY_FUNCTION__;
+	f.data = g723_slin_ex;
+	return &f;
+}
+
+static struct ast_translator_pvt *lintog723_new()
+{
+	struct g723_encoder_pvt *tmp;
+	tmp = malloc(sizeof(struct g723_encoder_pvt));
+	if (tmp) {
+		Init_Coder(&tmp->cod);
+	    /* Init Comfort Noise Functions */
+   		 if( UseVx ) {
+   	   		Init_Vad(&tmp->cod);
+        	Init_Cod_Cng(&tmp->cod);
+    	 }
+		tmp->tail = 0;
+	}
+	return (struct ast_translator_pvt *)tmp;
+}
+
+static struct ast_frame *g723tolin_frameout(struct ast_translator_pvt *pvt)
+{
+	struct g723_decoder_pvt *tmp = (struct g723_decoder_pvt *)pvt;
+	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("g723.example", O_WRONLY | O_CREAT | O_TRUNC, 0644);
+		write(fd, tmp->f.data, tmp->f.datalen);
+		close(fd);
+	} 		
+	samplefr++;
+	}
+#endif
+	return &tmp->f;	
+}
+
+static int g723tolin_framein(struct ast_translator_pvt *pvt, struct ast_frame *f)
+{
+	struct g723_decoder_pvt *tmp = (struct g723_decoder_pvt *)pvt;
+#ifdef  ANNEX_B
+	FLOAT tmpdata[Frame];
+	int x;
+#endif
+	/* Assuming there's space left, decode into the current buffer at
+	   the tail location */
+	if (tmp->tail + Frame < sizeof(tmp->buf)/2) {	
+#ifdef ANNEX_B
+		Decod(&tmp->dec, tmpdata, f->data, 0);
+		for (x=0;x<Frame;x++)
+			(tmp->buf + tmp->tail)[x] = tmpdata[x]; 
+#else
+		Decod(&tmp->dec, tmp->buf + tmp->tail, f->data, 0);
+#endif
+		tmp->tail+=Frame;
+	} else {
+		ast_log(LOG_WARNING, "Out of buffer space\n");
+		return -1;
+	}
+	return 0;
+}
+
+static int lintog723_framein(struct ast_translator_pvt *pvt, 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 */
+	struct g723_encoder_pvt *tmp = (struct g723_encoder_pvt *)pvt;
+	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");
+		return -1;
+	}
+	return 0;
+}
+
+static struct ast_frame *lintog723_frameout(struct ast_translator_pvt *pvt)
+{
+	struct g723_encoder_pvt *tmp = (struct g723_encoder_pvt *)pvt;
+#ifdef ANNEX_B
+	int x;
+	FLOAT tmpdata[Frame];
+#endif
+	/* We can't work on anything less than a frame in size */
+	if (tmp->tail < Frame)
+		return NULL;
+	/* Encode a frame of data */
+#ifdef ANNEX_B
+	for (x=0;x<Frame;x++)
+		tmpdata[x] = tmp->buf[x];
+	Coder(&tmp->cod, tmpdata, tmp->outbuf);
+#else
+	Coder(&tmp->cod, tmp->buf, tmp->outbuf);
+#endif
+	tmp->f.frametype = AST_FRAME_VOICE;
+	tmp->f.subclass = AST_FORMAT_G723_1;
+	/* Assume 8000 Hz */
+	tmp->f.timelen = 30;
+	tmp->f.mallocd = 0;
+	tmp->f.offset = AST_FRIENDLY_OFFSET;
+	tmp->f.src = __PRETTY_FUNCTION__;
+	tmp->f.data = tmp->outbuf;
+	switch(tmp->outbuf[0] & TYPE_MASK) {
+	case TYPE_MASK:
+	case TYPE_SILENCE:
+		tmp->f.datalen = 4;
+		break;
+	case TYPE_HIGH:
+		tmp->f.datalen = 24;
+		break;
+	case TYPE_LOW:
+		tmp->f.datalen = 20;
+		break;
+	default:
+		ast_log(LOG_WARNING, "Badly encoded frame (%d)\n", tmp->outbuf[0] & TYPE_MASK);
+	}
+	tmp->tail -= Frame;
+	/* Move the data at the end of the buffer to the front */
+	if (tmp->tail)
+		memmove(tmp->buf, tmp->buf + Frame, tmp->tail * 2);
+#if 0
+	/* Save to a g723 sample output file... */
+	{ 
+		static int fd = -1;
+		int delay = htonl(30);
+		short size;
+		if (fd < 0)
+			fd = open("trans.g723", O_WRONLY | O_CREAT | O_TRUNC, 0644);
+		if (fd < 0)
+			ast_log(LOG_WARNING, "Unable to create demo\n");
+		write(fd, &delay, 4);
+		size = htons(tmp->f.datalen);
+		write(fd, &size, 2);
+		write(fd, tmp->f.data, tmp->f.datalen);
+	}
+#endif
+	return &tmp->f;	
+}
+
+static void g723_destroy(struct ast_translator_pvt *pvt)
+{
+	free(pvt);
+}
+
+static struct ast_translator g723tolin =
+#ifdef ANNEX_B
+	{ "g723tolinb", 
+#else
+	{ "g723tolin", 
+#endif
+	   AST_FORMAT_G723_1, AST_FORMAT_SLINEAR,
+	   g723tolin_new,
+	   g723tolin_framein,
+	   g723tolin_frameout,
+	   g723_destroy,
+	   g723tolin_sample
+	   };
+
+static struct ast_translator lintog723 =
+#ifdef ANNEX_B
+	{ "lintog723b", 
+#else
+	{ "lintog723", 
+#endif
+	   AST_FORMAT_SLINEAR, AST_FORMAT_G723_1,
+	   lintog723_new,
+	   lintog723_framein,
+	   lintog723_frameout,
+	   g723_destroy,
+	   lintog723_sample
+	   };
+
+int unload_module(void)
+{
+	int res;
+	pthread_mutex_lock(&localuser_lock);
+	res = ast_unregister_translator(&lintog723);
+	if (!res)
+		res = ast_unregister_translator(&g723tolin);
+	if (localusecnt)
+		res = -1;
+	pthread_mutex_unlock(&localuser_lock);
+	return res;
+}
+
+int load_module(void)
+{
+	int res;
+	res=ast_register_translator(&g723tolin);
+	if (!res) 
+		res=ast_register_translator(&lintog723);
+	else
+		ast_unregister_translator(&g723tolin);
+	return res;
+}
+
+char *description(void)
+{
+	return tdesc;
+}
+
+int usecount(void)
+{
+	int res;
+	STANDARD_USECOUNT(res);
+	return res;
+}