1 files changed, 290 insertions, 0 deletions
diff --git a/1.4.23-rc4/codecs/codec_gsm.c b/1.4.23-rc4/codecs/codec_gsm.c
new file mode 100644
index 000000000..8a3749319
--- /dev/null
+++ b/1.4.23-rc4/codecs/codec_gsm.c
@@ -0,0 +1,290 @@
+/*
+ * Asterisk -- An open source telephony toolkit.
+ *
+ * The GSM code is from TOAST.  Copyright information for that package is available
+ * in the GSM directory.
+ *
+ * Copyright (C) 1999 - 2005, Digium, Inc.
+ *
+ * Mark Spencer <markster@digium.com>
+ *
+ * See http://www.asterisk.org for more information about
+ * the Asterisk project. Please do not directly contact
+ * any of the maintainers of this project for assistance;
+ * the project provides a web site, mailing lists and IRC
+ * channels for your use.
+ *
+ * This program is free software, distributed under the terms of
+ * the GNU General Public License Version 2. See the LICENSE file
+ * at the top of the source tree.
+ */
+
+/*! \file
+ *
+ * \brief Translate between signed linear and Global System for Mobile Communications (GSM)
+ *
+ * \ingroup codecs
+ */
+
+/*** MODULEINFO
+	<depend>gsm</depend>
+ ***/
+
+#include "asterisk.h"
+
+ASTERISK_FILE_VERSION(__FILE__, "$Revision$")
+
+#include <fcntl.h>
+#include <stdlib.h>
+#include <unistd.h>
+#include <netinet/in.h>
+#include <string.h>
+#include <stdio.h>
+
+#include "asterisk/lock.h"
+#include "asterisk/translate.h"
+#include "asterisk/config.h"
+#include "asterisk/options.h"
+#include "asterisk/module.h"
+#include "asterisk/logger.h"
+#include "asterisk/channel.h"
+#include "asterisk/utils.h"
+
+#ifdef HAVE_GSM_HEADER
+#include "gsm.h"
+#elif defined(HAVE_GSM_GSM_HEADER)
+#include <gsm/gsm.h>
+#endif
+
+#include "../formats/msgsm.h"
+
+/* Sample frame data */
+#include "slin_gsm_ex.h"
+#include "gsm_slin_ex.h"
+
+#define BUFFER_SAMPLES	8000
+#define GSM_SAMPLES	160
+#define	GSM_FRAME_LEN	33
+#define	MSGSM_FRAME_LEN	65
+
+struct gsm_translator_pvt {	/* both gsm2lin and lin2gsm */
+	gsm gsm;
+	int16_t buf[BUFFER_SAMPLES];	/* lin2gsm, temporary storage */
+};
+
+static int gsm_new(struct ast_trans_pvt *pvt)
+{
+	struct gsm_translator_pvt *tmp = pvt->pvt;
+	
+	return (tmp->gsm = gsm_create()) ? 0 : -1;
+}
+
+static struct ast_frame *lintogsm_sample(void)
+{
+	static struct ast_frame f;
+	f.frametype = AST_FRAME_VOICE;
+	f.subclass = AST_FORMAT_SLINEAR;
+	f.datalen = sizeof(slin_gsm_ex);
+	/* Assume 8000 Hz */
+	f.samples = sizeof(slin_gsm_ex)/2;
+	f.mallocd = 0;
+	f.offset = 0;
+	f.src = __PRETTY_FUNCTION__;
+	f.data = slin_gsm_ex;
+	return &f;
+}
+
+static struct ast_frame *gsmtolin_sample(void)
+{
+	static struct ast_frame f;
+	f.frametype = AST_FRAME_VOICE;
+	f.subclass = AST_FORMAT_GSM;
+	f.datalen = sizeof(gsm_slin_ex);
+	/* All frames are 20 ms long */
+	f.samples = GSM_SAMPLES;
+	f.mallocd = 0;
+	f.offset = 0;
+	f.src = __PRETTY_FUNCTION__;
+	f.data = gsm_slin_ex;
+	return &f;
+}
+
+/*! \brief decode and store in outbuf. */
+static int gsmtolin_framein(struct ast_trans_pvt *pvt, struct ast_frame *f)
+{
+	struct gsm_translator_pvt *tmp = pvt->pvt;
+	int x;
+	int16_t *dst = (int16_t *)pvt->outbuf;
+	/* guess format from frame len. 65 for MSGSM, 33 for regular GSM */
+	int flen = (f->datalen % MSGSM_FRAME_LEN == 0) ?
+		MSGSM_FRAME_LEN : GSM_FRAME_LEN;
+
+	for (x=0; x < f->datalen; x += flen) {
+		unsigned char data[2 * GSM_FRAME_LEN];
+		unsigned char *src;
+		int len;
+		if (flen == MSGSM_FRAME_LEN) {
+			len = 2*GSM_SAMPLES;
+			src = data;
+			/* Translate MSGSM format to Real GSM format before feeding in */
+			/* XXX what's the point here! we should just work
+			 * on the full format.
+			 */
+			conv65(f->data + x, data);
+		} else {
+			len = GSM_SAMPLES;
+			src = f->data + x;
+		}
+		/* XXX maybe we don't need to check */
+		if (pvt->samples + len > BUFFER_SAMPLES) {	
+			ast_log(LOG_WARNING, "Out of buffer space\n");
+			return -1;
+		}
+		if (gsm_decode(tmp->gsm, src, dst + pvt->samples)) {
+			ast_log(LOG_WARNING, "Invalid GSM data (1)\n");
+			return -1;
+		}
+		pvt->samples += GSM_SAMPLES;
+		pvt->datalen += 2 * GSM_SAMPLES;
+		if (flen == MSGSM_FRAME_LEN) {
+			if (gsm_decode(tmp->gsm, data + GSM_FRAME_LEN, dst + pvt->samples)) {
+				ast_log(LOG_WARNING, "Invalid GSM data (2)\n");
+				return -1;
+			}
+			pvt->samples += GSM_SAMPLES;
+			pvt->datalen += 2 * GSM_SAMPLES;
+		}
+	}
+	return 0;
+}
+
+/*! \brief store samples into working buffer for later decode */
+static int lintogsm_framein(struct ast_trans_pvt *pvt, struct ast_frame *f)
+{
+	struct gsm_translator_pvt *tmp = pvt->pvt;
+
+	/* 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 (pvt->samples + f->samples > BUFFER_SAMPLES) {
+		ast_log(LOG_WARNING, "Out of buffer space\n");
+		return -1;
+	}
+	memcpy(tmp->buf + pvt->samples, f->data, f->datalen);
+	pvt->samples += f->samples;
+	return 0;
+}
+
+/*! \brief encode and produce a frame */
+static struct ast_frame *lintogsm_frameout(struct ast_trans_pvt *pvt)
+{
+	struct gsm_translator_pvt *tmp = pvt->pvt;
+	int datalen = 0;
+	int samples = 0;
+
+	/* We can't work on anything less than a frame in size */
+	if (pvt->samples < GSM_SAMPLES)
+		return NULL;
+	while (pvt->samples >= GSM_SAMPLES) {
+		/* Encode a frame of data */
+		gsm_encode(tmp->gsm, tmp->buf + samples, (gsm_byte *) pvt->outbuf + datalen);
+		datalen += GSM_FRAME_LEN;
+		samples += GSM_SAMPLES;
+		pvt->samples -= GSM_SAMPLES;
+	}
+
+	/* Move the data at the end of the buffer to the front */
+	if (pvt->samples)
+		memmove(tmp->buf, tmp->buf + samples, pvt->samples * 2);
+
+	return ast_trans_frameout(pvt, datalen, samples);
+}
+
+static void gsm_destroy_stuff(struct ast_trans_pvt *pvt)
+{
+	struct gsm_translator_pvt *tmp = pvt->pvt;
+	if (tmp->gsm)
+		gsm_destroy(tmp->gsm);
+}
+
+static struct ast_translator gsmtolin = {
+	.name = "gsmtolin", 
+	.srcfmt = AST_FORMAT_GSM,
+	.dstfmt = AST_FORMAT_SLINEAR,
+	.newpvt = gsm_new,
+	.framein = gsmtolin_framein,
+	.destroy = gsm_destroy_stuff,
+	.sample = gsmtolin_sample,
+	.buffer_samples = BUFFER_SAMPLES,
+	.buf_size = BUFFER_SAMPLES * 2,
+	.desc_size = sizeof (struct gsm_translator_pvt ),
+	.plc_samples = GSM_SAMPLES,
+};
+
+static struct ast_translator lintogsm = {
+	.name = "lintogsm", 
+	.srcfmt = AST_FORMAT_SLINEAR,
+	.dstfmt = AST_FORMAT_GSM,
+	.newpvt = gsm_new,
+	.framein = lintogsm_framein,
+	.frameout = lintogsm_frameout,
+	.destroy = gsm_destroy_stuff,
+	.sample = lintogsm_sample,
+	.desc_size = sizeof (struct gsm_translator_pvt ),
+	.buf_size = (BUFFER_SAMPLES * GSM_FRAME_LEN + GSM_SAMPLES - 1)/GSM_SAMPLES,
+};
+
+
+static void parse_config(void)
+{
+	struct ast_variable *var;
+	struct ast_config *cfg = ast_config_load("codecs.conf");
+	if (!cfg)
+		return;
+	for (var = ast_variable_browse(cfg, "plc"); var; var = var->next) {
+	       if (!strcasecmp(var->name, "genericplc")) {
+		       gsmtolin.useplc = ast_true(var->value) ? 1 : 0;
+		       if (option_verbose > 2)
+			       ast_verbose(VERBOSE_PREFIX_3 "codec_gsm: %susing generic PLC\n", gsmtolin.useplc ? "" : "not ");
+	       }
+	}
+	ast_config_destroy(cfg);
+}
+
+/*! \brief standard module glue */
+static int reload(void)
+{
+	parse_config();
+	return 0;
+}
+
+static int unload_module(void)
+{
+	int res;
+
+	res = ast_unregister_translator(&lintogsm);
+	if (!res)
+		res = ast_unregister_translator(&gsmtolin);
+
+	return res;
+}
+
+static int load_module(void)
+{
+	int res;
+
+	parse_config();
+	res = ast_register_translator(&gsmtolin);
+	if (!res) 
+		res=ast_register_translator(&lintogsm);
+	else
+		ast_unregister_translator(&gsmtolin);
+
+	return res;
+}
+
+AST_MODULE_INFO(ASTERISK_GPL_KEY, AST_MODFLAG_DEFAULT, "GSM Coder/Decoder",
+		.load = load_module,
+		.unload = unload_module,
+		.reload = reload,
+	       );