Add HeaderExtensions to RtpParameters

Bug: webrtc:7580
Change-Id: I4fcf3e8bc4975a6b2baa6f24a17c254d2bf521d9
Reviewed-on: https://webrtc-review.googlesource.com/78288
Commit-Queue: Florent Castelli <orphis@webrtc.org>
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Reviewed-by: Anders Carlsson <andersc@webrtc.org>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23584}

pc/peerconnection_rtp_unittest.cc

diff --git a/pc/peerconnection_rtp_unittest.cc b/pc/peerconnection_rtp_unittest.cc
index f4a174b..586189e 100644
--- a/pc/peerconnection_rtp_unittest.cc
+++ b/pc/peerconnection_rtp_unittest.cc
@@ -566,6 +566,34 @@
   }
 }
 
+TEST_P(PeerConnectionRtpTest, AudioGetParametersHasHeaderExtensions) {
+  auto caller = CreatePeerConnection();
+  auto callee = CreatePeerConnection();
+  auto sender = caller->AddAudioTrack("audio_track");
+  ASSERT_TRUE(caller->ExchangeOfferAnswerWith(callee.get()));
+
+  ASSERT_GT(caller->pc()->GetSenders().size(), 0u);
+  EXPECT_GT(sender->GetParameters().header_extensions.size(), 0u);
+
+  ASSERT_GT(callee->pc()->GetReceivers().size(), 0u);
+  auto receiver = callee->pc()->GetReceivers()[0];
+  EXPECT_GT(receiver->GetParameters().header_extensions.size(), 0u);
+}
+
+TEST_P(PeerConnectionRtpTest, VideoGetParametersHasHeaderExtensions) {
+  auto caller = CreatePeerConnection();
+  auto callee = CreatePeerConnection();
+  auto sender = caller->AddVideoTrack("video_track");
+  ASSERT_TRUE(caller->ExchangeOfferAnswerWith(callee.get()));
+
+  ASSERT_GT(caller->pc()->GetSenders().size(), 0u);
+  EXPECT_GT(sender->GetParameters().header_extensions.size(), 0u);
+
+  ASSERT_GT(callee->pc()->GetReceivers().size(), 0u);
+  auto receiver = callee->pc()->GetReceivers()[0];
+  EXPECT_GT(receiver->GetParameters().header_extensions.size(), 0u);
+}
+
 // Invokes SetRemoteDescription() twice in a row without synchronizing the two
 // calls and examine the state of the peer connection inside the callbacks to
 // ensure that the second call does not occur prematurely, contaminating the

pc/rtpsender.cc

diff --git a/pc/rtpsender.cc b/pc/rtpsender.cc
index b7e48cd..c7fe83f 100644
--- a/pc/rtpsender.cc
+++ b/pc/rtpsender.cc
@@ -68,7 +68,7 @@
 // Returns true if any RtpParameters member that isn't implemented contains a
 // value.
 bool UnimplementedRtpParameterHasValue(const RtpParameters& parameters) {
-  if (!parameters.mid.empty() || !parameters.header_extensions.empty() ||
+  if (!parameters.mid.empty() ||
       parameters.degradation_preference != DegradationPreference::BALANCED) {
     return true;
   }

pc/rtpsenderreceiver_unittest.cc

diff --git a/pc/rtpsenderreceiver_unittest.cc b/pc/rtpsenderreceiver_unittest.cc
index be0e975..1575e96 100644
--- a/pc/rtpsenderreceiver_unittest.cc
+++ b/pc/rtpsenderreceiver_unittest.cc
@@ -681,11 +681,6 @@
             audio_rtp_sender_->SetParameters(params).type());
   params = audio_rtp_sender_->GetParameters();
 
-  params.header_extensions.emplace_back();
-  EXPECT_EQ(RTCErrorType::UNSUPPORTED_PARAMETER,
-            audio_rtp_sender_->SetParameters(params).type());
-  params = audio_rtp_sender_->GetParameters();
-
   ASSERT_EQ(DegradationPreference::BALANCED, params.degradation_preference);
   params.degradation_preference = DegradationPreference::MAINTAIN_FRAMERATE;
   EXPECT_EQ(RTCErrorType::UNSUPPORTED_PARAMETER,
@@ -879,11 +874,6 @@
             video_rtp_sender_->SetParameters(params).type());
   params = video_rtp_sender_->GetParameters();
 
-  params.header_extensions.emplace_back();
-  EXPECT_EQ(RTCErrorType::UNSUPPORTED_PARAMETER,
-            video_rtp_sender_->SetParameters(params).type());
-  params = video_rtp_sender_->GetParameters();
-
   ASSERT_EQ(DegradationPreference::BALANCED, params.degradation_preference);
   params.degradation_preference = DegradationPreference::MAINTAIN_FRAMERATE;
   EXPECT_EQ(RTCErrorType::UNSUPPORTED_PARAMETER,