commit 8b47ea459e849e2d9d13c2792a8fafe7fc726f64
author Dan Tan <dwtan@google.com> Wed Nov 23 01:02:02 2022 -0800
committer WebRTC LUCI CQ <webrtc-scoped@luci-project-accounts.iam.gserviceaccount.com> Tue Nov 29 18:18:57 2022 +0000
tree c1530a761a4a2fa03936013a404ff0fbeb5c41e2
parent 41a83571707a574bbe3da32f564b963b8092d59b

Fixed timestamp_offset for RtpSenderEgress during initialization and SetRtpState call

The constructor and SetRtpState calls for ModuleRtpRtcpImpl2 class fail to propagate the RTP timestamp offset of RtpSender class to RtpSenderEgress class. This results in wrong RTP timestamps being propagated in LossNotification messages.

Change-Id: I1d293289a4815de29d9dd15208bb7fd1a682be82
Bug: webrtc:14719
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/284824
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Commit-Queue: Dan Tan <dwtan@google.com>
Cr-Commit-Position: refs/heads/main@{#38768}

modules/rtp_rtcp/source/rtp_rtcp_impl2.cc

diff --git a/modules/rtp_rtcp/source/rtp_rtcp_impl2.cc b/modules/rtp_rtcp/source/rtp_rtcp_impl2.cc
index 4329a42..ecc05ca 100644
--- a/modules/rtp_rtcp/source/rtp_rtcp_impl2.cc
+++ b/modules/rtp_rtcp/source/rtp_rtcp_impl2.cc
@@ -89,6 +89,8 @@
     // Make sure rtcp sender use same timestamp offset as rtp sender.
     rtcp_sender_.SetTimestampOffset(
         rtp_sender_->packet_generator.TimestampOffset());
+    rtp_sender_->packet_sender.SetTimestampOffset(
+        rtp_sender_->packet_generator.TimestampOffset());
   }
 
   // Set default packet size limit.
@@ -186,6 +188,7 @@
   rtp_sender_->packet_generator.SetRtpState(rtp_state);
   rtp_sender_->sequencer.SetRtpState(rtp_state);
   rtcp_sender_.SetTimestampOffset(rtp_state.start_timestamp);
+  rtp_sender_->packet_sender.SetTimestampOffset(rtp_state.start_timestamp);
 }
 
 void ModuleRtpRtcpImpl2::SetRtxState(const RtpState& rtp_state) {

modules/rtp_rtcp/source/rtp_rtcp_impl2_unittest.cc

diff --git a/modules/rtp_rtcp/source/rtp_rtcp_impl2_unittest.cc b/modules/rtp_rtcp/source/rtp_rtcp_impl2_unittest.cc
index 4dece66..f5c6311 100644
--- a/modules/rtp_rtcp/source/rtp_rtcp_impl2_unittest.cc
+++ b/modules/rtp_rtcp/source/rtp_rtcp_impl2_unittest.cc
@@ -709,6 +709,36 @@
   EXPECT_EQ(sender_.transport_.NumRtcpSent(), 2u);
 }
 
+TEST_P(RtpRtcpImpl2Test, RtpSenderEgressTimestampOffset) {
+  // RTP timestamp offset not explicitly set, default to random value.
+  uint16_t seqno = sender_.impl_->GetRtpState().sequence_number;
+  uint32_t media_rtp_ts = 1001;
+  uint32_t rtp_ts = media_rtp_ts + sender_.impl_->StartTimestamp();
+  EXPECT_TRUE(SendFrame(&sender_, sender_video_.get(), kBaseLayerTid, rtp_ts,
+                        /*capture_time_ms=*/0));
+  AdvanceTime(kOneWayNetworkDelay);
+  EXPECT_THAT(
+      sender_.impl_->GetSentRtpPacketInfos(std::vector<uint16_t>{seqno}),
+      ElementsAre(Field(&RtpSequenceNumberMap::Info::timestamp, media_rtp_ts)));
+
+  RtpState saved_rtp_state = sender_.impl_->GetRtpState();
+
+  // Change RTP timestamp offset.
+  sender_.impl_->SetStartTimestamp(2000);
+
+  // Restores RtpState and make sure the old timestamp offset is in place.
+  sender_.impl_->SetRtpState(saved_rtp_state);
+  seqno = sender_.impl_->GetRtpState().sequence_number;
+  media_rtp_ts = 1031;
+  rtp_ts = media_rtp_ts + sender_.impl_->StartTimestamp();
+  EXPECT_TRUE(SendFrame(&sender_, sender_video_.get(), kBaseLayerTid, rtp_ts,
+                        /*capture_time_ms=*/0));
+  AdvanceTime(kOneWayNetworkDelay);
+  EXPECT_THAT(
+      sender_.impl_->GetSentRtpPacketInfos(std::vector<uint16_t>{seqno}),
+      ElementsAre(Field(&RtpSequenceNumberMap::Info::timestamp, media_rtp_ts)));
+}
+
 TEST_P(RtpRtcpImpl2Test, StoresPacketInfoForSentPackets) {
   const uint32_t kStartTimestamp = 1u;
   SetUp();