[asterisk-commits] oej: branch oej/rana-dtmf-duration-1.8 r389843 - /team/oej/rana-dtmf-duration...

[SVN commits to the Asterisk project]

Author: oej
Date: Tue May 28 04:40:26 2013
New Revision: 389843

URL: http://svnview.digium.com/svn/asterisk?view=rev&rev=389843
Log:
Fixing some issues found when testing

- Repeated BEGIN transmissions should not add to the duration. That's a bug.
- If we receive 10 ms DTMF we should buffer, since we only send 20 ms.
  If the 20ms hard-coding is a bug is another issue, Asterisk DTMF doesn't play 
  well with other packetization. This needs to be fixed at some point.
- Reset received duration at start of new DTMF

Modified:
    team/oej/rana-dtmf-duration-1.8/res/res_rtp_asterisk.c

Modified: team/oej/rana-dtmf-duration-1.8/res/res_rtp_asterisk.c
URL: http://svnview.digium.com/svn/asterisk/team/oej/rana-dtmf-duration-1.8/res/res_rtp_asterisk.c?view=diff&rev=389843&r1=389842&r2=389843
==============================================================================
--- team/oej/rana-dtmf-duration-1.8/res/res_rtp_asterisk.c (original)
+++ team/oej/rana-dtmf-duration-1.8/res/res_rtp_asterisk.c Tue May 28 04:40:26 2013
@@ -722,6 +722,7 @@
 
 	rtp->dtmfmute = ast_tvadd(ast_tvnow(), ast_tv(0, 500000));
 	rtp->send_duration = 160;
+	rtp->received_duration = 160;
 	rtp->lastdigitts = rtp->lastts + rtp->send_duration;
 
 	/* Create the actual packet that we will be sending */
@@ -744,7 +745,7 @@
 				    payload, rtp->seqno, rtp->lastdigitts, res - hdrlen);
 		}
 		rtp->seqno++;
-		rtp->send_duration += 160;	/* OEJ - check what's going on here. */
+		//rtp->send_duration += 160;	/* OEJ - check what's going on here. */
 		
 		rtpheader[0] = htonl((2 << 30) | (payload << 16) | (rtp->seqno));
 	}
@@ -754,7 +755,7 @@
 	rtp->send_digit = digit;
 	rtp->send_payload = payload;
 
-	ast_log(LOG_DEBUG, "DEBUG DTMF BEGIN - Digit %d send-digit %d\n", digit, rtp->send_digit);
+	ast_debug(4, "DEBUG DTMF BEGIN - Digit %d send-digit %d\n", digit, rtp->send_digit);
 
 	return 0;
 }
@@ -819,20 +820,27 @@
 		return -1;
 	}
 
-	ast_debug(4, "---- Send duration %d Received duration %d Endflag %d Send-digit %d\n", rtp->send_duration, rtp->received_duration, rtp->send_endflag, rtp->send_digit);
 
 	/*! \todo XXX This code assumes 160 samples, which is for 20 ms of 8000 samples
 		we need to calculate this based on the current sample rate and the rtp 
 		stream packetization. Please help me figure this out :-)
 	 */
-	if (rtp->received_duration == 0 || rtp->send_duration + 160 <= rtp->received_duration) {
+	if (!rtp->send_endflag && rtp->send_duration + 160 > rtp->received_duration) {
+		/* We need to wait with sending this continue, as we're sending 160 frames */
+		ast_debug(4, "---- Send duration %d Received duration %d - Skipping this continue frame until we have a proper 20 ms/160 samples to send\n", rtp->send_duration, rtp->received_duration);
+		return -1;
+	}
+	if (rtp->received_duration == 0 || rtp->send_duration + 160 < rtp->received_duration) {
+		ast_debug(3, "---- Adding 160 samples before sending : (previous values) Send duration %d Received duration %d\n", rtp->send_duration, rtp->received_duration);
 		rtp->send_duration += 160;
-	} else if (rtp->send_endflag) {
-		ast_log(LOG_DEBUG, "---- Send duration %d Received duration %d - sending END packet\n", rtp->send_duration, rtp->received_duration);
+	} 
+	if (rtp->send_endflag) {
+		ast_debug(4, "---- Send duration %d Received duration %d - sending END packet\n", rtp->send_duration, rtp->received_duration);
 		/* We are done, ready to send end flag */
 		rtp->send_endflag = 0;
 		return ast_rtp_dtmf_end_with_duration(instance, 0, rtp->received_duration);
 	}
+	ast_debug(4, "---- Send duration %d Received duration %d Endflag %d Send-digit %d\n", rtp->send_duration, rtp->received_duration, rtp->send_endflag, rtp->send_digit);
 	/* Actually create the packet we will be sending */
 	rtpheader[0] = htonl((2 << 30) | (rtp->send_payload << 16) | (rtp->seqno));
 	rtpheader[1] = htonl(rtp->lastdigitts);
@@ -856,6 +864,7 @@
 	/* And now we increment some values for the next time we swing by */
 	rtp->seqno++;
 	rtp->send_duration += 160;	/* Again assuming 20 ms packetization and 8000 samples */
+	ast_debug(4, "---- Adding 160 samples after sending : Send duration %d Received duration %d\n", rtp->send_duration, rtp->received_duration);
 
 	return 0;
 }
@@ -884,7 +893,7 @@
 		/* We still have to send DTMF continuation, because otherwise we will end prematurely. Set end flag to indicate
 		   that we will have to end ourselves when we're done with the actual duration
 		 */
-		ast_log(LOG_DEBUG, "---- Send duration %d Received duration %d - Avoiding sending END packet\n", rtp->send_duration, rtp->received_duration);
+		ast_debug(4, "---- Send duration %d Received duration %d - Avoiding sending END packet\n", rtp->send_duration, rtp->received_duration);
 		rtp->send_endflag = 1;
 		return ast_rtp_dtmf_cont(instance);
 	}
@@ -941,6 +950,8 @@
 
 	/* Oh and we can't forget to turn off the stuff that says we are sending DTMF */
 	rtp->lastts += rtp->send_duration;
+
+cleanup:
 	rtp->sending_dtmf = DTMF_NOT_SENDING;
 	rtp->send_digit = 0;
 
@@ -1628,7 +1639,7 @@
 		return &ast_null_frame;
 	}
 	ast_debug(1, "Creating %s DTMF Frame: %d (%c), at %s\n",
-		type == AST_FRAME_DTMF_END ? "END" : "BEGIN",
+		type == AST_FRAME_DTMF_END ? "END" : "BEGIN/CONT",
 		rtp->resp, rtp->resp,
 		ast_sockaddr_stringify(&remote_address));
 	if (rtp->resp == 'X') {
@@ -1957,7 +1968,7 @@
 		if ((i + length) > packetwords) {
 			if (rtpdebug || option_debug) {
 				/* Because of rtpdebug, this can't be ast_debug() */
-				ast_log(LOG_DEBUG, "RTCP Read too short\n");
+				ast_debug(1, "RTCP Read too short\n");
 			}
 			return &ast_null_frame;
 		}