[asterisk-commits] mjordan: branch 11 r385637 - in /branches/11: ./ res/res_rtp_multicast.c

[SVN commits to the Asterisk project]

Author: mjordan
Date: Sat Apr 13 22:00:27 2013
New Revision: 385637

URL: http://svnview.digium.com/svn/asterisk?view=rev&rev=385637
Log:
Calculate the timestamp for outbound RTP if we don't have timing information

This patch calculates the timestamp for outbound RTP when we don't have timing
information. This uses the same approach in res_rtp_asterisk. Thanks to both
Pietro and Tzafrir for providing patches.

(closes issue ASTERISK-19883)
Reported by: Giacomo Trovato
Tested by: Pietro Bertera, Tzafrir Cohen
patches:
  rtp-timestamp-1.8.patch uploaded by tzafrir (License 5035)
  rtp-timestamp.patch uploaded by pbertera (License 5943)
........

Merged revisions 385636 from http://svn.asterisk.org/svn/asterisk/branches/1.8

Modified:
    branches/11/   (props changed)
    branches/11/res/res_rtp_multicast.c

Propchange: branches/11/
------------------------------------------------------------------------------
Binary property 'branch-1.8-merged' - no diff available.

Modified: branches/11/res/res_rtp_multicast.c
URL: http://svnview.digium.com/svn/asterisk/branches/11/res/res_rtp_multicast.c?view=diff&rev=385637&r1=385636&r2=385637
==============================================================================
--- branches/11/res/res_rtp_multicast.c (original)
+++ branches/11/res/res_rtp_multicast.c Sat Apr 13 22:00:27 2013
@@ -90,6 +90,8 @@
 	unsigned int ssrc;
 	/*! Sequence number, used when creating/sending the RTP packet */
 	uint16_t seqno;
+	unsigned int lastts;	
+	struct timeval txcore;
 };
 
 /* Forward Declarations */
@@ -140,6 +142,30 @@
 	return 0;
 }
 
+static int rtp_get_rate(struct ast_format *format)
+{
+        return (format->id == AST_FORMAT_G722) ? 8000 : ast_format_rate(format);
+}
+
+static unsigned int calc_txstamp(struct multicast_rtp *rtp, struct timeval *delivery)
+{
+        struct timeval t;
+        long ms;
+
+        if (ast_tvzero(rtp->txcore)) {
+                rtp->txcore = ast_tvnow();
+                rtp->txcore.tv_usec -= rtp->txcore.tv_usec % 20000;
+        }
+
+        t = (delivery && !ast_tvzero(*delivery)) ? *delivery : ast_tvnow();
+        if ((ms = ast_tvdiff_ms(t, rtp->txcore)) < 0) {
+                ms = 0;
+        }
+        rtp->txcore = t;
+
+        return (unsigned int) ms;
+}
+
 /*! \brief Helper function which populates a control packet with useful information and sends it */
 static int multicast_send_control_packet(struct ast_rtp_instance *instance, struct multicast_rtp *multicast, int command)
 {
@@ -210,6 +236,8 @@
 	struct ast_sockaddr remote_address;
 	int hdrlen = 12, res = 0, codec;
 	unsigned char *rtpheader;
+	unsigned int ms = calc_txstamp(multicast, &frame->delivery);
+	int rate = rtp_get_rate(&frame->subclass.format) / 1000;
 
 	/* We only accept audio, nothing else */
 	if (frame->frametype != AST_FRAME_VOICE) {
@@ -225,12 +253,21 @@
 	if (frame->offset < hdrlen) {
 		f = ast_frdup(frame);
 	}
-
+	
+	/* Calucate last TS */
+	multicast->lastts = multicast->lastts + ms * rate;
+	
 	/* Construct an RTP header for our packet */
 	rtpheader = (unsigned char *)(f->data.ptr - hdrlen);
 	put_unaligned_uint32(rtpheader, htonl((2 << 30) | (codec << 16) | (multicast->seqno)));
-	put_unaligned_uint32(rtpheader + 4, htonl(f->ts * 8));
-	put_unaligned_uint32(rtpheader + 8, htonl(multicast->ssrc));
+	put_unaligned_uint32(rtpheader + 4, htonl(multicast->lastts));
+	
+	if (ast_test_flag(f, AST_FRFLAG_HAS_TIMING_INFO)) {
+		put_unaligned_uint32(rtpheader + 4, htonl(f->ts * 8));
+	} 
+	else {
+		put_unaligned_uint32(rtpheader + 8, htonl(multicast->ssrc));
+	}
 
 	/* Increment sequence number and wrap to 0 if it overflows 16 bits. */
 	multicast->seqno = 0xFFFF & (multicast->seqno + 1);