[asterisk-commits] rtp: Preserve timestamps on video frames. (asterisk[master])

[SVN commits to the Asterisk project]

Anonymous Coward #1000019 has submitted this change and it was merged.

Change subject: rtp: Preserve timestamps on video frames.
......................................................................


rtp: Preserve timestamps on video frames.

Currently when receiving video over RTP we store only
a calculated samples on the frame. When starting the video
it can take some time for this calculation to actually yield
a value as it requires constant changing timestamps. As well
if a video frame passes over multiple RTP packets this calculation
will fail as the timestamp is the same as the previous RTP
packet and the number of samples calculated will be 0.

This change preserves the timestamp on the frame and allows
it to pass through the core. When sending the video this timestamp
is used instead of a new one being calculated.

ASTERISK-26367 #close

Change-Id: Iba8179fb5c14c9443aee4baf670d2185da3ecfbd
---
M main/codec_builtin.c
M res/res_rtp_asterisk.c
2 files changed, 8 insertions(+), 0 deletions(-)

Approvals:
  Mark Michelson: Looks good to me, approved
  Richard Mudgett: Looks good to me, but someone else must approve
  Anonymous Coward #1000019: Verified



diff --git a/main/codec_builtin.c b/main/codec_builtin.c
index facb5a8..1f2cb92 100644
--- a/main/codec_builtin.c
+++ b/main/codec_builtin.c
@@ -758,36 +758,42 @@
 	.name = "h261",
 	.description = "H.261 video",
 	.type = AST_MEDIA_TYPE_VIDEO,
+	.sample_rate = 1000,
 };
 
 static struct ast_codec h263 = {
 	.name = "h263",
 	.description = "H.263 video",
 	.type = AST_MEDIA_TYPE_VIDEO,
+	.sample_rate = 1000,
 };
 
 static struct ast_codec h263p = {
 	.name = "h263p",
 	.description = "H.263+ video",
 	.type = AST_MEDIA_TYPE_VIDEO,
+	.sample_rate = 1000,
 };
 
 static struct ast_codec h264 = {
 	.name = "h264",
 	.description = "H.264 video",
 	.type = AST_MEDIA_TYPE_VIDEO,
+	.sample_rate = 1000,
 };
 
 static struct ast_codec mpeg4 = {
 	.name = "mpeg4",
 	.description = "MPEG4 video",
 	.type = AST_MEDIA_TYPE_VIDEO,
+	.sample_rate = 1000,
 };
 
 static struct ast_codec vp8 = {
 	.name = "vp8",
 	.description = "VP8 video",
 	.type = AST_MEDIA_TYPE_VIDEO,
+	.sample_rate = 1000,
 };
 
 static struct ast_codec t140red = {
diff --git a/res/res_rtp_asterisk.c b/res/res_rtp_asterisk.c
index a8d4a90..d6eea3c 100644
--- a/res/res_rtp_asterisk.c
+++ b/res/res_rtp_asterisk.c
@@ -4763,6 +4763,8 @@
 		/* Video -- samples is # of samples vs. 90000 */
 		if (!rtp->lastividtimestamp)
 			rtp->lastividtimestamp = timestamp;
+		ast_set_flag(&rtp->f, AST_FRFLAG_HAS_TIMING_INFO);
+		rtp->f.ts = timestamp / (rtp_get_rate(rtp->f.subclass.format) / 1000);
 		rtp->f.samples = timestamp - rtp->lastividtimestamp;
 		rtp->lastividtimestamp = timestamp;
 		rtp->f.delivery.tv_sec = 0;

-- 
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Gerrit-MessageType: merged
Gerrit-Change-Id: Iba8179fb5c14c9443aee4baf670d2185da3ecfbd
Gerrit-PatchSet: 2
Gerrit-Project: asterisk
Gerrit-Branch: master
Gerrit-Owner: Joshua Colp <jcolp at digium.com>
Gerrit-Reviewer: Anonymous Coward #1000019
Gerrit-Reviewer: Mark Michelson <mmichelson at digium.com>
Gerrit-Reviewer: Richard Mudgett <rmudgett at digium.com>