[Asterisk-cvs] asterisk rtp.c,1.84,1.85

[markster at lists.digium.com]

Update of /usr/cvsroot/asterisk
In directory mongoose.digium.com:/tmp/cvs-serv15880

Modified Files:
	rtp.c 
Log Message:
Repair offer/answer model (bug #2293), initial CNG work for new frametype


Index: rtp.c
===================================================================
RCS file: /usr/cvsroot/asterisk/rtp.c,v
retrieving revision 1.84
retrieving revision 1.85
diff -u -d -r1.84 -r1.85
--- rtp.c	7 Aug 2004 14:22:09 -0000	1.84
+++ rtp.c	26 Aug 2004 04:56:26 -0000	1.85
@@ -5,7 +5,7 @@
  * 	Supports RTP and RTCP with Symmetric RTP support for NAT
  * 	traversal
  * 
- * Copyright (C) 1999, Mark Spencer
+ * Copyright (C) 1999-2004, Digium, Inc.
  *
  * Mark Spencer <markster at digium.com>
  *
@@ -99,6 +99,7 @@
     int rtp_lookup_code_cache_isAstFormat;	/* a cache for the result of rtp_lookup_code(): */
     int rtp_lookup_code_cache_code;
     int rtp_lookup_code_cache_result;
+    int rtp_offered_from_local;
 	struct ast_rtcp *rtcp;
 };
 
@@ -281,43 +282,28 @@
 	/* Convert comfort noise into audio with various codecs.  Unfortunately this doesn't
 	   totally help us out becuase we don't have an engine to keep it going and we are not
 	   guaranteed to have it every 20ms or anything */
-#if 0
-	printf("RFC3389: %d bytes, format is %d\n", len, rtp->lastrxformat);
+#if 1
+	printf("RFC3389: %d bytes, level %d...\n", len, rtp->lastrxformat);
 #endif	
 	if (!(rtp->flags & FLAG_3389_WARNING)) {
 		ast_log(LOG_NOTICE, "RFC3389 support incomplete.  Turn off on client if possible\n");
 		rtp->flags |= FLAG_3389_WARNING;
 	}
-	if (!rtp->lastrxformat)
-		return 	NULL;
-	switch(rtp->lastrxformat) {
-	case AST_FORMAT_ULAW:
-		rtp->f.frametype = AST_FRAME_VOICE;
-		rtp->f.subclass = AST_FORMAT_ULAW;
-		rtp->f.datalen = 160;
-		rtp->f.samples = 160;
-		memset(rtp->f.data, 0x7f, rtp->f.datalen);
-		f = &rtp->f;
-		break;
-	case AST_FORMAT_ALAW:
-		rtp->f.frametype = AST_FRAME_VOICE;
-		rtp->f.subclass = AST_FORMAT_ALAW;
-		rtp->f.datalen = 160;
-		rtp->f.samples = 160;
-		memset(rtp->f.data, 0x7e, rtp->f.datalen); /* XXX Is this right? XXX */
-		f = &rtp->f;
-		break;
-	case AST_FORMAT_SLINEAR:
-		rtp->f.frametype = AST_FRAME_VOICE;
-		rtp->f.subclass = AST_FORMAT_SLINEAR;
-		rtp->f.datalen = 320;
-		rtp->f.samples = 160;
-		memset(rtp->f.data, 0x00, rtp->f.datalen);
-		f = &rtp->f;
-		break;
-	default:
-		ast_log(LOG_NOTICE, "Don't know how to handle RFC3389 for receive codec %d\n", rtp->lastrxformat);
+	/* Must have at least one byte */
+	if (!len)
+		return NULL;
+	if (len < 24) {
+		rtp->f.datalen = len - 1;
+		memcpy(rtp->f.data, data + 1, len - 1);
+	} else {
+		rtp->f.datalen = 0;
 	}
+	rtp->f.frametype = AST_FRAME_CNG;
+	rtp->f.subclass = data[0] & 0x7f;
+	rtp->f.datalen = len - 1;
+	rtp->f.samples = 0;
+	rtp->f.delivery.tv_usec = rtp->f.delivery.tv_sec = 0;
+	f = &rtp->f;
 	return f;
 }
 
@@ -587,7 +573,7 @@
   {{1, AST_FORMAT_SLINEAR}, "audio", "L16"},
   {{1, AST_FORMAT_LPC10}, "audio", "LPC"},
   {{1, AST_FORMAT_G729A}, "audio", "G729"},
-  {{1, AST_FORMAT_SPEEX}, "audio", "SPEEX"},
+  {{1, AST_FORMAT_SPEEX}, "audio", "speex"},
   {{1, AST_FORMAT_ILBC}, "audio", "iLBC"},
   {{0, AST_RTP_DTMF}, "audio", "telephone-event"},
   {{0, AST_RTP_CISCO_DTMF}, "audio", "cisco-telephone-event"},
@@ -698,19 +684,27 @@
   }
 }
 
+void ast_rtp_offered_from_local(struct ast_rtp* rtp, int local) {
+  if (rtp)
+    rtp->rtp_offered_from_local = local;
+  else
+    ast_log(LOG_WARNING, "rtp structure is null\n");
+}
+
 struct rtpPayloadType ast_rtp_lookup_pt(struct ast_rtp* rtp, int pt) 
 {
   struct rtpPayloadType result;
 
+  result.isAstFormat = result.code = 0;
   if (pt < 0 || pt > MAX_RTP_PT) {
-    result.isAstFormat = result.code = 0;
     return result; /* bogus payload type */
   }
   /* Start with the negotiated codecs */
-  result = rtp->current_RTP_PT[pt];
+  if (!rtp->rtp_offered_from_local)
+    result = rtp->current_RTP_PT[pt];
   /* If it doesn't exist, check our static RTP type list, just in case */
   if (!result.code) 
-  	result = static_RTP_PT[pt];
+    result = static_RTP_PT[pt];
   return result;
 }