Forward the SignalFirstPacketReceived to RtpReceiver.

webrtc/api/peerconnection_unittest.cc

 /*
  *  Copyright 2012 The WebRTC project authors. All Rights Reserved.
  *
  *  Use of this source code is governed by a BSD-style license
  *  that can be found in the LICENSE file in the root of the source
  *  tree. An additional intellectual property rights grant can be found
  *  in the file PATENTS.  All contributing project authors may
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
@@ skipping 128 matching lines @@
                                  std::string& msg) = 0;
   virtual void ReceiveIceMessage(const std::string& sdp_mid,
                                  int sdp_mline_index,
                                  const std::string& msg) = 0;
 
  protected:
   SignalingMessageReceiver() {}
   virtual ~SignalingMessageReceiver() {}
 };
 
+class MockRtpReceiverObserver : public webrtc::RtpReceiverObserverInterface {
+ public:
+  MockRtpReceiverObserver()
+      : is_first_audio_packet_recevied_(false),
+        is_first_video_packet_recevied_(false),
+        kind_("unset") {}
+
+  void onFirstPacketReceived(cricket::MediaType media_type) override {

[pthatcher2, 2016/05/20 20:23:13]

OnFirstPacketReceived

+    if (media_type == cricket::MEDIA_TYPE_AUDIO) {
+      is_first_audio_packet_recevied_ = true;
+    } else if (media_type == cricket::MEDIA_TYPE_VIDEO) {
+      is_first_video_packet_recevied_ = true;
+    }
+  }
+
+  void SetKind(std::string kind) { kind_ = kind; }
+
+  std::string kind() { return kind_; }
+
+  bool IsFirstAudioPacketReceived() { return is_first_audio_packet_recevied_; }
+
+  bool IsFirstVideoPacketReceived() { return is_first_video_packet_recevied_; }
+
+  virtual ~MockRtpReceiverObserver() {}
+
+ private:
+  bool is_first_audio_packet_recevied_;
+  bool is_first_video_packet_recevied_;
+  std::string kind_;
+};
+
 class PeerConnectionTestClient : public webrtc::PeerConnectionObserver,
                                  public SignalingMessageReceiver,
                                  public ObserverInterface {
  public:
   static PeerConnectionTestClient* CreateClientWithDtlsIdentityStore(
       const std::string& id,
       const MediaConstraintsInterface* constraints,
       const PeerConnectionFactory::Options* options,
       std::unique_ptr<webrtc::DtlsIdentityStoreInterface> dtls_identity_store,
       bool prefer_constraint_apis,
@@ skipping 611 matching lines @@
   }
 
   webrtc::PeerConnectionInterface::IceConnectionState ice_connection_state() {
     return pc()->ice_connection_state();
   }
 
   webrtc::PeerConnectionInterface::IceGatheringState ice_gathering_state() {
     return pc()->ice_gathering_state();
   }
 
+  std::vector<rtc::scoped_refptr<MockRtpReceiverObserver>>
+  RtpReceiverObservers() {

[pthatcher2, 2016/05/20 20:23:13]

rtp_receiver_observers()

[zhihuang1, 2016/05/24 01:01:44]

Done.

+    return rtp_receiver_observers_;
+  }
+
  private:
   class DummyDtmfObserver : public DtmfSenderObserverInterface {
    public:
     DummyDtmfObserver() : completed_(false) {}
 
     // Implements DtmfSenderObserverInterface.
     void OnToneChange(const std::string& tone) override {
       tones_.push_back(tone);
       if (tone.empty()) {
         completed_ = true;
@@ skipping 70 matching lines @@
 
   void HandleIncomingOffer(const std::string& msg) {
     LOG(INFO) << id_ << "HandleIncomingOffer ";
     if (NumberOfLocalMediaStreams() == 0 && auto_add_stream_) {
       // If we are not sending any streams ourselves it is time to add some.
       AddMediaStream(true, true);
     }
     std::unique_ptr<SessionDescriptionInterface> desc(
         webrtc::CreateSessionDescription("offer", msg, nullptr));
     EXPECT_TRUE(DoSetRemoteDescription(desc.release()));
+    // Register the RtpReceiverObserver after receivers are created.
+    DoRegisterRtpReceiverObservers();

[pthatcher2, 2016/05/20 20:23:13]

Why not just RegisterRtpReceiverObservers()?

[zhihuang1, 2016/05/24 01:01:44]

Done.

     std::unique_ptr<SessionDescriptionInterface> answer;
     EXPECT_TRUE(DoCreateAnswer(&answer));
     std::string sdp;
     EXPECT_TRUE(answer->ToString(&sdp));
     EXPECT_TRUE(DoSetLocalDescription(answer.release()));
     if (signaling_message_receiver_) {
       signaling_message_receiver_->ReceiveSdpMessage(
           webrtc::SessionDescriptionInterface::kAnswer, sdp);
     }
   }
@@ skipping 53 matching lines @@
     // ProcessMessages waits at least 1ms but processes all messages before
     // returning. Since this test is synchronous and send messages to the remote
     // peer whenever a callback is invoked, this can lead to messages being
     // sent to the remote peer in the wrong order.
     // TODO(perkj): Find a way to check the result without risking that the
     // order of sent messages are changed. Ex- by posting all messages that are
     // sent to the remote peer.
     return true;
   }
 
+  void DoRegisterRtpReceiverObservers() {
+    for (auto receiver : pc()->GetReceivers()) {
+      rtc::scoped_refptr<MockRtpReceiverObserver> observer(
+          new rtc::RefCountedObject<MockRtpReceiverObserver>());
+      observer->SetKind(receiver->track()->kind());
+      receiver->RegisterRtpReceiverObserver(observer);
+      rtp_receiver_observers_.push_back(observer);
+    }
+  }
+
   bool DoSetRemoteDescription(SessionDescriptionInterface* desc) {
     rtc::scoped_refptr<MockSetSessionDescriptionObserver>
         observer(new rtc::RefCountedObject<
             MockSetSessionDescriptionObserver>());
     LOG(INFO) << id_ << "SetRemoteDescription ";
     pc()->SetRemoteDescription(observer, desc);
     EXPECT_TRUE_WAIT(observer->called(), kMaxWaitMs);
     return observer->result();
   }
 
@@ skipping 63 matching lines @@
   bool remove_bundle_ =
       false;  // True if bundle should be removed in received SDP.
   bool remove_sdes_ =
       false;  // True if a=crypto should be removed in received SDP.
   // |remove_cvo_| is true if extension urn:3gpp:video-orientation should be
   // removed in the received SDP.
   bool remove_cvo_ = false;
 
   rtc::scoped_refptr<DataChannelInterface> data_channel_;
   std::unique_ptr<MockDataChannelObserver> data_observer_;
+
+  std::vector<rtc::scoped_refptr<MockRtpReceiverObserver>>
+      rtp_receiver_observers_;
 };
 
 class P2PTestConductor : public testing::Test {
  public:
   P2PTestConductor()
       : network_thread_(rtc::Thread::CreateWithSocketServer()),
         worker_thread_(rtc::Thread::Create()),
         pss_(new rtc::PhysicalSocketServer),
         ss_(new rtc::VirtualSocketServer(pss_.get())),
         ss_scope_(ss_.get()) {
@@ skipping 269 matching lines @@
 
   // Set the |receiving_client_| to the |client| passed in and return the
   // original |receiving_client_|.
   PeerConnectionTestClient* set_receiving_client(
       PeerConnectionTestClient* client) {
     PeerConnectionTestClient* old = receiving_client_.release();
     receiving_client_.reset(client);
     return old;
   }
 
+  // Only check the callee side because only callee can guarantee to register
+  // the observer before sending the data.

[pthatcher2, 2016/05/20 20:23:13]

Here's a simple solution:  whenever an observer is added after the event has
fired, tell the observer that the evern has fired by calling
OnFirstPacketReceived.

[zhihuang1, 2016/05/24 01:01:44]

This is a nice and easy solution.

+  void CheckRtpReceiverObserverAreCalled() {
+    for (auto observer : receiving_client_->RtpReceiverObservers()) {
+      if (observer->kind() == webrtc::MediaStreamTrackInterface::kAudioKind) {
+        EXPECT_TRUE_WAIT(observer->IsFirstAudioPacketReceived(),
+                         kMaxWaitForFramesMs);
+        EXPECT_FALSE(observer->IsFirstVideoPacketReceived());
+      } else if (observer->kind() ==
+                 webrtc::MediaStreamTrackInterface::kVideoKind) {
+        EXPECT_FALSE(observer->IsFirstAudioPacketReceived());
+        EXPECT_TRUE_WAIT(observer->IsFirstVideoPacketReceived(),
+                         kMaxWaitForFramesMs);
+      } else {
+        EXPECT_TRUE(false);

[pthatcher2, 2016/05/20 20:23:13]

I think you can add a message to the EXPECT, like "Unknown RtpReceiver kind"

[zhihuang1, 2016/05/24 01:01:44]

Done.

+      }
+    }
+  }
+
  private:
   // |worker_thread_| is used by both |initiating_client_| and
   // |receiving_client_|. Must be destroyed last.
   std::unique_ptr<rtc::Thread> network_thread_;
   std::unique_ptr<rtc::Thread> worker_thread_;
   std::unique_ptr<rtc::PhysicalSocketServer> pss_;
   std::unique_ptr<rtc::VirtualSocketServer> ss_;
   rtc::SocketServerScope ss_scope_;
   std::unique_ptr<PeerConnectionTestClient> initiating_client_;
   std::unique_ptr<PeerConnectionTestClient> receiving_client_;
   bool prefer_constraint_apis_ = true;
 };
 
 // Disable for TSan v2, see
 // https://code.google.com/p/webrtc/issues/detail?id=1205 for details.
 #if !defined(THREAD_SANITIZER)
 
+TEST_F(P2PTestConductor, TestRtpReceiverObserverCallbackFunction) {
+  ASSERT_TRUE(CreateTestClients());
+  LocalP2PTest();
+  CheckRtpReceiverObserverAreCalled();

[pthatcher2, 2016/05/20 20:23:13]

If it's only called once, just put the code inline and don't bother with a
separate function.

[zhihuang1, 2016/05/24 01:01:44]

I will make it a helper function to check both the caller and callee side.

+}
+
 // This test sets up a Jsep call between two parties and test Dtmf.
 // TODO(holmer): Disabled due to sometimes crashing on buildbots.
 // See issue webrtc/2378.
 TEST_F(P2PTestConductor, DISABLED_LocalP2PTestDtmf) {
   ASSERT_TRUE(CreateTestClients());
   LocalP2PTest();
   VerifyDtmf();
 }
 
 // This test sets up a Jsep call between two parties and test that we can get a
@@ skipping 869 matching lines @@
   server.urls.push_back("turn:hostname2");
   servers.push_back(server);
   EXPECT_TRUE(webrtc::ParseIceServers(servers, &stun_servers_, &turn_servers_));
   EXPECT_EQ(2U, turn_servers_.size());
   EXPECT_NE(turn_servers_[0].priority, turn_servers_[1].priority);
 }
 
 #endif // if !defined(THREAD_SANITIZER)
 
 }  // namespace