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(cricket::MediaType media_type)
+      : expected_media_type_(media_type) {}
+
+  void OnFirstPacketReceived(cricket::MediaType media_type) override {
+    ASSERT_EQ(expected_media_type_, media_type);
+    first_packet_received_ = true;
+  }
+
+  bool first_packet_received() { return first_packet_received_; }
+
+  virtual ~MockRtpReceiverObserver() {}
+
+ private:
+  bool first_packet_received_ = false;
+  cricket::MediaType expected_media_type_;
+};
+
 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<rtc::RTCCertificateGeneratorInterface> cert_generator,
       bool prefer_constraint_apis,
@@ skipping 614 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<std::unique_ptr<MockRtpReceiverObserver>> const&
+  rtp_receiver_observers() {
+    return rtp_receiver_observers_;
+  }
+
+  void SetRtpReceiverObservers() {
+    rtp_receiver_observers_.clear();
+    for (auto receiver : pc()->GetReceivers()) {
+      std::unique_ptr<MockRtpReceiverObserver> observer(
+          new MockRtpReceiverObserver(receiver->media_type()));
+      receiver->SetObserver(observer.get());
+      rtp_receiver_observers_.push_back(std::move(observer));
+    }
+  }
+
  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()));
+    // Set the RtpReceiverObserver after receivers are created.
+    SetRtpReceiverObservers();
     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);
     }
   }
 
   void HandleIncomingAnswer(const std::string& msg) {
     LOG(INFO) << id_ << "HandleIncomingAnswer";
     std::unique_ptr<SessionDescriptionInterface> desc(
         webrtc::CreateSessionDescription("answer", msg, nullptr));
     EXPECT_TRUE(DoSetRemoteDescription(desc.release()));
+    // Set the RtpReceiverObserver after receivers are created.
+    SetRtpReceiverObservers();
   }
 
   bool DoCreateOfferAnswer(std::unique_ptr<SessionDescriptionInterface>* desc,
                            bool offer) {
     rtc::scoped_refptr<MockCreateSessionDescriptionObserver>
         observer(new rtc::RefCountedObject<
             MockCreateSessionDescriptionObserver>());
     if (prefer_constraint_apis_) {
       if (offer) {
         pc()->CreateOffer(observer, &offer_answer_constraints_);
@@ skipping 120 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<std::unique_ptr<MockRtpReceiverObserver>> rtp_receiver_observers_;
 };
 
 class P2PTestConductor : public testing::Test {
  public:
   P2PTestConductor()
       : pss_(new rtc::PhysicalSocketServer),
         ss_(new rtc::VirtualSocketServer(pss_.get())),
         network_thread_(new rtc::Thread(ss_.get())),
         worker_thread_(rtc::Thread::Create()) {
     RTC_CHECK(network_thread_->Start());
@@ skipping 268 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;
   }
 
+  void CheckRtpReceiverObserverAreCalled(PeerConnectionTestClient* client) {

[pthatcher1, 2016/06/13 23:27:27]

This should probably be:

EXPECT_TRUE_WAIT(std::all_of(client->rtp_receiver_observers(),
[](std::unique_ptr<MockRtpReceiverObserver> observer) { return
observer->first_packet_received(); }), kMaxWaitForFramesMs);

To avoid waiting many times.

(I'm not sure if that compiles, but you get the idea :).


But then, it might be cleaner to make the method thus:

bool AllObserversReceived(const
std::vector<std::unique_ptr<MockRtpReceiverObserver>>& observers) {
  return std::all_of(client->rtp_receiver_observers(),
[](std::unique_ptr<MockRtpReceiverObserver> observer) { return
observer->first_packet_received(); });
}

And then later:

EXPECT_TRUE_WAIT(AllObserversReceived(initializing_client()->rtp_receiver_observers()),
kMaxWaitForFramesMs);
EXPECT_TRUE_WAIT(AllObserversReceived(receiving_client()->rtp_receiver_observers()),
kMaxWaitForFramesMs);

+    for (auto const& observer : client->rtp_receiver_observers()) {
+      EXPECT_TRUE_WAIT(observer->first_packet_received(), kMaxWaitForFramesMs);
+    }
+  }
+
  private:
   // |ss_| is used by |network_thread_| so it must be destroyed later.
   std::unique_ptr<rtc::PhysicalSocketServer> pss_;
   std::unique_ptr<rtc::VirtualSocketServer> ss_;
   // |network_thread_| and |worker_thread_| are used by both
   // |initiating_client_| and |receiving_client_| so they must be destroyed
   // later.
   std::unique_ptr<rtc::Thread> network_thread_;
   std::unique_ptr<rtc::Thread> worker_thread_;
   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(initializing_client());
+  CheckRtpReceiverObserverAreCalled(receiving_client());
+}
+
+// The observers are expected to fire the signal even if they are set after the
+// first packet is received.
+TEST_F(P2PTestConductor, TestSetRtpReceiverObserverAfterFirstPacketIsReceived) {
+  ASSERT_TRUE(CreateTestClients());
+  LocalP2PTest();
+  // Reset the RtpReceiverObservers.
+  initializing_client()->SetRtpReceiverObservers();
+  receiving_client()->SetRtpReceiverObservers();
+  CheckRtpReceiverObserverAreCalled(initializing_client());
+  CheckRtpReceiverObserverAreCalled(receiving_client());
+}
+
 // 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