Adds a test for an one-way media PeerConnection.

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 397 matching lines @@
                                                     &source_constraints);
     std::string label = stream_label + kVideoTrackLabelBase;
     return peer_connection_factory_->CreateVideoTrack(label, source);
   }
 
   DataChannelInterface* data_channel() { return data_channel_; }
   const MockDataChannelObserver* data_observer() const {
     return data_observer_.get();
   }
 
-  webrtc::PeerConnectionInterface* pc() { return peer_connection_.get(); }
+  webrtc::PeerConnectionInterface* pc() const { return peer_connection_.get(); }
 
   void StopVideoCapturers() {
     for (std::vector<cricket::VideoCapturer*>::iterator it =
              video_capturers_.begin();
          it != video_capturers_.end(); ++it) {
       (*it)->Stop();
     }
   }
 
   bool AudioFramesReceivedCheck(int number_of_frames) const {
     return number_of_frames <= fake_audio_capture_module_->frames_received();
   }
 
   int audio_frames_received() const {
     return fake_audio_capture_module_->frames_received();
   }
 
   bool VideoFramesReceivedCheck(int number_of_frames) {
     if (video_decoder_factory_enabled_) {
       const std::vector<FakeWebRtcVideoDecoder*>& decoders
           = fake_video_decoder_factory_->decoders();
       if (decoders.empty()) {
         return number_of_frames <= 0;
       }
-
+      // Note - this checks that EACH decoder has the requisite number
+      // of frames. The video_frames_received() function sums them.
       for (FakeWebRtcVideoDecoder* decoder : decoders) {
         if (number_of_frames > decoder->GetNumFramesReceived()) {
           return false;
         }
       }
       return true;
     } else {
       if (fake_video_renderers_.empty()) {
         return number_of_frames <= 0;
       }
@@ skipping 199 matching lines @@
     return fake_video_renderers_.empty() ? 1 :
         fake_video_renderers_.begin()->second->height();
   }
 
   size_t number_of_remote_streams() {
     if (!pc())
       return 0;
     return pc()->remote_streams()->count();
   }
 
-  StreamCollectionInterface* remote_streams() {
+  StreamCollectionInterface* remote_streams() const {
     if (!pc()) {
       ADD_FAILURE();
       return nullptr;
     }
     return pc()->remote_streams();
   }
 
   StreamCollectionInterface* local_streams() {
     if (!pc()) {
       ADD_FAILURE();
@@ skipping 238 matching lines @@
   P2PTestConductor()
       : pss_(new rtc::PhysicalSocketServer),
         ss_(new rtc::VirtualSocketServer(pss_.get())),
         ss_scope_(ss_.get()) {}
 
   bool SessionActive() {
     return initiating_client_->SessionActive() &&
            receiving_client_->SessionActive();
   }
 
-  // Return true if the number of frames provided have been received or it is
-  // known that that will never occur (e.g. no frames will be sent or
-  // captured).
-  bool FramesNotPending(int audio_frames_to_receive,
-                        int video_frames_to_receive) {
-    return VideoFramesReceivedCheck(video_frames_to_receive) &&
-        AudioFramesReceivedCheck(audio_frames_to_receive);
+  // Return true if the number of frames provided have been received
+  // on the video and audio tracks provided.
+  bool FramesHaveArrived(int audio_frames_to_receive,
+                         int video_frames_to_receive) {
+    bool all_good = true;
+    // Note: We assume that all tracks contain audio and video,
+    // unless one of the parties has can_receive_video() = false.
+    if (initiating_client_->local_streams()->count() > 0) {
+      if (receiving_client_->can_receive_audio()) {
+        all_good &= receiving_client_->AudioFramesReceivedCheck(
+            audio_frames_to_receive);
+      }
+      if (receiving_client_->can_receive_video() &&
+          initiating_client_->can_receive_video()) {
+        all_good &= receiving_client_->VideoFramesReceivedCheck(
+            audio_frames_to_receive);

[perkj_webrtc, 2016/03/09 16:56:25]

video_frames_to_receive

[hta-webrtc, 2016/03/09 17:50:56]

Done.

+      }
+    }
+    if (receiving_client_->local_streams()->count() > 0) {
+      if (initiating_client_->can_receive_audio() &&
+          receiving_client_->can_receive_audio()) {

[perkj_webrtc, 2016/03/09 16:56:25]

please change to 
if (receiving_client_->local_streams()->count() > 0) && if
(receiving_client_->local_streams()->at(0)->GetAudioTracks().size() > 0)

instead.

[hta-webrtc, 2016/03/09 17:50:56]

Restructured.

+        all_good &= initiating_client_->AudioFramesReceivedCheck(
+            audio_frames_to_receive);
+      }
+      if (initiating_client_->can_receive_video() &&
+          receiving_client_->can_receive_video()) {
+        all_good &= initiating_client_->VideoFramesReceivedCheck(
+            audio_frames_to_receive);

[perkj_webrtc, 2016/03/09 16:56:25]

video_frames_to_receive

[hta-webrtc, 2016/03/09 17:50:56]

Done.

+      }
+    }
+    return all_good;
   }
-  bool AudioFramesReceivedCheck(int frames_received) {
-    return initiating_client_->AudioFramesReceivedCheck(frames_received) &&
-        receiving_client_->AudioFramesReceivedCheck(frames_received);
-  }
-  bool VideoFramesReceivedCheck(int frames_received) {
-    return initiating_client_->VideoFramesReceivedCheck(frames_received) &&
-        receiving_client_->VideoFramesReceivedCheck(frames_received);
-  }
+
   void VerifyDtmf() {
     initiating_client_->VerifyDtmf();
     receiving_client_->VerifyDtmf();
   }
 
   void TestUpdateOfferWithRejectedContent() {
     // Renegotiate, rejecting the video m-line.
     initiating_client_->Negotiate(true, false);
     ASSERT_TRUE_WAIT(SessionActive(), kMaxWaitForActivationMs);
 
@@ skipping 84 matching lines @@
     if (initiating_client_->NumberOfLocalMediaStreams() == 0) {
       initiating_client_->AddMediaStream(true, true);
     }
     initiating_client_->Negotiate();
     // Assert true is used here since next tests are guaranteed to fail and
     // would eat up 5 seconds.
     ASSERT_TRUE_WAIT(SessionActive(), kMaxWaitForActivationMs);
     VerifySessionDescriptions();
 
     int audio_frame_count = kEndAudioFrameCount;
+    int video_frame_count = kEndVideoFrameCount;
     // TODO(ronghuawu): Add test to cover the case of sendonly and recvonly.

[perkj_webrtc, 2016/03/09 16:56:25]

please remove this todo

[hta-webrtc, 2016/03/09 17:50:56]

Don't think it's right to remove. We still have no tests for sendonly/recvonly
in SDP.

-    if (!initiating_client_->can_receive_audio() ||
-        !receiving_client_->can_receive_audio()) {
-      audio_frame_count = -1;
-    }
-    int video_frame_count = kEndVideoFrameCount;
-    if (!initiating_client_->can_receive_video() ||
-        !receiving_client_->can_receive_video()) {
-      video_frame_count = -1;
+
+    if ((!initiating_client_->can_receive_audio() &&
+         !initiating_client_->can_receive_video()) ||
+        (!receiving_client_->can_receive_audio() &&
+         !receiving_client_->can_receive_video())) {
+      // Neither audio nor video will flow, so connections won't be
+      // established. There's nothing more to check.
+      // TODO(hta): Check connection if there's a data channel.
+      return;
     }
 
-    if (audio_frame_count != -1 || video_frame_count != -1) {
-      // Audio or video is expected to flow, so both clients should reach the
-      // Connected state, and the offerer (ICE controller) should proceed to
-      // Completed.
-      // Note: These tests have been observed to fail under heavy load at
-      // shorter timeouts, so they may be flaky.
-      EXPECT_EQ_WAIT(
-          webrtc::PeerConnectionInterface::kIceConnectionCompleted,
-          initiating_client_->ice_connection_state(),
-          kMaxWaitForFramesMs);
-      EXPECT_EQ_WAIT(
-          webrtc::PeerConnectionInterface::kIceConnectionConnected,
-          receiving_client_->ice_connection_state(),
-          kMaxWaitForFramesMs);
-    }
+    // Audio or video is expected to flow, so both clients should reach the
+    // Connected state, and the offerer (ICE controller) should proceed to
+    // Completed.
+    // Note: These tests have been observed to fail under heavy load at
+    // shorter timeouts, so they may be flaky.
+    EXPECT_EQ_WAIT(webrtc::PeerConnectionInterface::kIceConnectionCompleted,
+                   initiating_client_->ice_connection_state(),
+                   kMaxWaitForFramesMs);
+    EXPECT_EQ_WAIT(webrtc::PeerConnectionInterface::kIceConnectionConnected,
+                   receiving_client_->ice_connection_state(),
+                   kMaxWaitForFramesMs);
 
-    if (initiating_client_->can_receive_audio() ||
-        initiating_client_->can_receive_video()) {
-      // The initiating client can receive media, so it must produce candidates
-      // that will serve as destinations for that media.
-      // TODO(bemasc): Understand why the state is not already Complete here, as
-      // seems to be the case for the receiving client. This may indicate a bug
-      // in the ICE gathering system.
-      EXPECT_NE(webrtc::PeerConnectionInterface::kIceGatheringNew,
-                initiating_client_->ice_gathering_state());
-    }
-    if (receiving_client_->can_receive_audio() ||
-        receiving_client_->can_receive_video()) {
-      EXPECT_EQ_WAIT(webrtc::PeerConnectionInterface::kIceGatheringComplete,
-                     receiving_client_->ice_gathering_state(),
-                     kMaxWaitForFramesMs);
-    }
+    // The ICE gathering state should end up in kIceGatheringComplete,
+    // but there's a bug that prevents this at the moment, and the state
+    // machine is being updated by the WEBRTC WG.
+    // TODO(hta): Update this check when spec revisions finish.
+    EXPECT_NE(webrtc::PeerConnectionInterface::kIceGatheringNew,
+              initiating_client_->ice_gathering_state());
+    EXPECT_EQ_WAIT(webrtc::PeerConnectionInterface::kIceGatheringComplete,
+                   receiving_client_->ice_gathering_state(),
+                   kMaxWaitForFramesMs);
 
-    EXPECT_TRUE_WAIT(FramesNotPending(audio_frame_count, video_frame_count),
+    // Check that the expected number of frames have arrived.
+    EXPECT_TRUE_WAIT(FramesHaveArrived(audio_frame_count, video_frame_count),
                      kMaxWaitForFramesMs);
   }
 
   void SetupAndVerifyDtlsCall() {
     MAYBE_SKIP_TEST(rtc::SSLStreamAdapter::HaveDtlsSrtp);
     FakeConstraints setup_constraints;
     setup_constraints.AddMandatory(MediaConstraintsInterface::kEnableDtlsSrtp,
                                    true);
     ASSERT_TRUE(CreateTestClients(&setup_constraints, &setup_constraints));
     LocalP2PTest();
@@ skipping 109 matching lines @@
   LocalP2PTest();
   VerifyRenderedSize(1280, 720);
 }
 
 // This test sets up a call between two endpoints that are configured to use
 // DTLS key agreement. As a result, DTLS is negotiated and used for transport.
 TEST_F(P2PTestConductor, LocalP2PTestDtls) {
   SetupAndVerifyDtlsCall();
 }
 
+// This test sets up an one-way call, with media only from initiator to
+// responder.
+TEST_F(P2PTestConductor, OneWayMediaCall) {
+  ASSERT_TRUE(CreateTestClients());
+  receiving_client()->set_auto_add_stream(false);
+  LocalP2PTest();
+}
+
 // This test sets up a audio call initially and then upgrades to audio/video,
 // using DTLS.
 TEST_F(P2PTestConductor, LocalP2PTestDtlsRenegotiate) {
   MAYBE_SKIP_TEST(rtc::SSLStreamAdapter::HaveDtlsSrtp);
   FakeConstraints setup_constraints;
   setup_constraints.AddMandatory(MediaConstraintsInterface::kEnableDtlsSrtp,
                                  true);
   ASSERT_TRUE(CreateTestClients(&setup_constraints, &setup_constraints));
   receiving_client()->SetReceiveAudioVideo(true, false);
   LocalP2PTest();
@@ skipping 586 matching lines @@
                  initializing_client()->ice_connection_state(),
                  kMaxWaitForFramesMs);
   EXPECT_EQ_WAIT(webrtc::PeerConnectionInterface::kIceConnectionConnected,
                  receiving_client()->ice_connection_state(),
                  kMaxWaitForFramesMs);
   // Now set the tracks, and expect frames to immediately start flowing.
   EXPECT_TRUE(
       audio_sender->SetTrack(initializing_client()->CreateLocalAudioTrack("")));
   EXPECT_TRUE(
       video_sender->SetTrack(initializing_client()->CreateLocalVideoTrack("")));
-  EXPECT_TRUE_WAIT(FramesNotPending(kEndAudioFrameCount, kEndVideoFrameCount),
+  EXPECT_TRUE_WAIT(FramesHaveArrived(kEndAudioFrameCount, kEndVideoFrameCount),
                    kMaxWaitForFramesMs);
 }
 
 class IceServerParsingTest : public testing::Test {
  public:
   // Convenience for parsing a single URL.
   bool ParseUrl(const std::string& url) {
     return ParseUrl(url, std::string(), std::string());
   }
 
@@ skipping 173 matching lines @@
   PeerConnectionInterface::IceServer server;
   server.urls.push_back("turn:hostname");
   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)