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authormarkster <markster@f38db490-d61c-443f-a65b-d21fe96a405b>1999-12-05 01:40:43 +0000
committermarkster <markster@f38db490-d61c-443f-a65b-d21fe96a405b>1999-12-05 01:40:43 +0000
commit09fff5e6354c8603d3ca6df32ac705852425a8af (patch)
tree486272d58c5e3fe45aa8bdd8b36b4020883e26fc
parent08659117c8215fd1a57eee36844182e575a53c0e (diff)
Version 0.1.0 from FTP
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@87 f38db490-d61c-443f-a65b-d21fe96a405b
-rwxr-xr-xBUGS9
-rwxr-xr-xcodecs/Makefile72
-rwxr-xr-xcodecs/codec_g723_1.c358
3 files changed, 439 insertions, 0 deletions
diff --git a/BUGS b/BUGS
new file mode 100755
index 000000000..04fbe3e48
--- /dev/null
+++ b/BUGS
@@ -0,0 +1,9 @@
+* 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.
diff --git a/codecs/Makefile b/codecs/Makefile
new file mode 100755
index 000000000..8dc01b4b1
--- /dev/null
+++ b/codecs/Makefile
@@ -0,0 +1,72 @@
+#
+# 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
diff --git a/codecs/codec_g723_1.c b/codecs/codec_g723_1.c
new file mode 100755
index 000000000..d33ba8783
--- /dev/null
+++ b/codecs/codec_g723_1.c
@@ -0,0 +1,358 @@
+/*
+ * 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;
+}