[Asterisk-code-review] res rtp multicast: Use consistent timestamps when possible (asterisk[13])

[Sean Bright]

Sean Bright has uploaded a new change for review. ( https://gerrit.asterisk.org/5762 )

Change subject: res_rtp_multicast: Use consistent timestamps when possible
......................................................................

res_rtp_multicast: Use consistent timestamps when possible

When a frame destined for a MulticastRTP channel does not have timing
information (such as when an 'originate' is done), we generate the RTP
timestamps ourselves without regard to the number of samples we are
about to send.

Instead, use the same method as res_rtp_asterisk and 'predict' a
timestamp given the number of samples. If the difference between the
timestamp that we generate and the one we predict is within a specific
threshold, use the predicted timestamp so that we end up with timestamps
that are consistent with the number of samples we are actually sending.

Change-Id: I2bf0db3541b1573043330421cbb114ff0f22ec1f
---
M res/res_rtp_multicast.c
1 file changed, 28 insertions(+), 10 deletions(-)


  git pull ssh://gerrit.asterisk.org:29418/asterisk refs/changes/62/5762/1

diff --git a/res/res_rtp_multicast.c b/res/res_rtp_multicast.c
index f1a80e4..1a8de63 100644
--- a/res/res_rtp_multicast.c
+++ b/res/res_rtp_multicast.c
@@ -101,6 +101,8 @@
 	struct ast_smoother *smoother;
 };
 
+#define MAX_TIMESTAMP_SKEW 640
+
 enum {
 	OPT_CODEC = (1 << 0),
 	OPT_LOOP =  (1 << 1),
@@ -417,21 +419,37 @@
 	unsigned char *rtpheader;
 	struct ast_sockaddr remote_address = { {0,} };
 	int rate = rtp_get_rate(frame->subclass.format) / 1000;
-	int hdrlen = 12;
+	int hdrlen = 12, mark = 0;
 
-	/* Calculate last TS */
-	multicast->lastts = multicast->lastts + ms * rate;
+	if (ast_format_cmp(frame->subclass.format, ast_format_g722) == AST_FORMAT_CMP_EQUAL) {
+		frame->samples /= 2;
+	}
+
+	if (ast_test_flag(frame, AST_FRFLAG_HAS_TIMING_INFO)) {
+		multicast->lastts = frame->ts * rate;
+	} else {
+		/* Try to predict what our timestamp should be */
+		int pred = multicast->lastts + frame->samples;
+
+		/* Calculate last TS */
+		multicast->lastts = multicast->lastts + ms * rate;
+		if (ast_tvzero(frame->delivery)) {
+			int delta = abs((int) multicast->lastts - pred);
+			if (delta < MAX_TIMESTAMP_SKEW) {
+				multicast->lastts = pred;
+			} else {
+				ast_debug(3, "Difference is %d, ms is %u\n", delta, ms);
+				mark = 1;
+			}
+		}
+
+	}
 
 	/* Construct an RTP header for our packet */
 	rtpheader = (unsigned char *)(frame->data.ptr - hdrlen);
-	put_unaligned_uint32(rtpheader, htonl((2 << 30) | (codec << 16) | (multicast->seqno)));
 
-	if (ast_test_flag(frame, AST_FRFLAG_HAS_TIMING_INFO)) {
-		put_unaligned_uint32(rtpheader + 4, htonl(frame->ts * 8));
-	} else {
-		put_unaligned_uint32(rtpheader + 4, htonl(multicast->lastts));
-	}
-
+	put_unaligned_uint32(rtpheader, htonl((2 << 30) | (codec << 16) | (multicast->seqno) | (mark << 23)));
+	put_unaligned_uint32(rtpheader + 4, htonl(multicast->lastts));
 	put_unaligned_uint32(rtpheader + 8, htonl(multicast->ssrc));
 
 	/* Increment sequence number and wrap to 0 if it overflows 16 bits. */

-- 
To view, visit https://gerrit.asterisk.org/5762
To unsubscribe, visit https://gerrit.asterisk.org/settings

Gerrit-MessageType: newchange
Gerrit-Change-Id: I2bf0db3541b1573043330421cbb114ff0f22ec1f
Gerrit-PatchSet: 1
Gerrit-Project: asterisk
Gerrit-Branch: 13
Gerrit-Owner: Sean Bright <sean.bright at gmail.com>