Adding the ability to create an RtpSender without a track.

talk/app/webrtc/peerconnection_unittest.cc

 /*
  * libjingle
  * Copyright 2012 Google Inc.
  *
  * Redistribution and use in source and binary forms, with or without
  * modification, are permitted provided that the following conditions are met:
  *
  *  1. Redistributions of source code must retain the above copyright notice,
  *     this list of conditions and the following disclaimer.
  *  2. Redistributions in binary form must reproduce the above copyright notice,
@@ skipping 78 matching lines @@
 using webrtc::StreamCollectionInterface;
 
 static const int kMaxWaitMs = 10000;
 // Disable for TSan v2, see
 // https://code.google.com/p/webrtc/issues/detail?id=1205 for details.
 // This declaration is also #ifdef'd as it causes uninitialized-variable
 // warnings.
 #if !defined(THREAD_SANITIZER)
 static const int kMaxWaitForStatsMs = 3000;
 #endif
+static const int kMaxWaitForActivationMs = 5000;
 static const int kMaxWaitForFramesMs = 10000;
 static const int kEndAudioFrameCount = 3;
 static const int kEndVideoFrameCount = 3;
 
 static const char kStreamLabelBase[] = "stream_label";
 static const char kVideoTrackLabelBase[] = "video_track";
 static const char kAudioTrackLabelBase[] = "audio_track";
 static const char kDataChannelLabel[] = "data_channel";
 
 // Disable for TSan v2, see
@@ skipping 134 matching lines @@
   }
 
   void AddMediaStream(bool audio, bool video) {
     std::string stream_label =
         kStreamLabelBase +
         rtc::ToString<int>(static_cast<int>(pc()->local_streams()->count()));
     rtc::scoped_refptr<webrtc::MediaStreamInterface> stream =
         peer_connection_factory_->CreateLocalMediaStream(stream_label);
 
     if (audio && can_receive_audio()) {
-      FakeConstraints constraints;
-      // Disable highpass filter so that we can get all the test audio frames.
-      constraints.AddMandatory(
-          MediaConstraintsInterface::kHighpassFilter, false);
-      rtc::scoped_refptr<webrtc::AudioSourceInterface> source =
-          peer_connection_factory_->CreateAudioSource(&constraints);
-      // TODO(perkj): Test audio source when it is implemented. Currently audio
-      // always use the default input.
-      std::string label = stream_label + kAudioTrackLabelBase;
-      rtc::scoped_refptr<webrtc::AudioTrackInterface> audio_track(
-          peer_connection_factory_->CreateAudioTrack(label, source));
-      stream->AddTrack(audio_track);
+      stream->AddTrack(CreateLocalAudioTrack(stream_label));
     }
     if (video && can_receive_video()) {
       stream->AddTrack(CreateLocalVideoTrack(stream_label));
     }
 
     EXPECT_TRUE(pc()->AddStream(stream));
   }
 
   size_t NumberOfLocalMediaStreams() { return pc()->local_streams()->count(); }
 
@@ skipping 75 matching lines @@
     data_channel_ = data_channel;
     data_observer_.reset(new MockDataChannelObserver(data_channel));
   }
 
   void CreateDataChannel() {
     data_channel_ = pc()->CreateDataChannel(kDataChannelLabel, nullptr);
     ASSERT_TRUE(data_channel_.get() != nullptr);
     data_observer_.reset(new MockDataChannelObserver(data_channel_));
   }
 
+  rtc::scoped_refptr<webrtc::AudioTrackInterface> CreateLocalAudioTrack(
+      const std::string& stream_label) {
+    FakeConstraints constraints;
+    // Disable highpass filter so that we can get all the test audio frames.
+    constraints.AddMandatory(MediaConstraintsInterface::kHighpassFilter, false);
+    rtc::scoped_refptr<webrtc::AudioSourceInterface> source =
+        peer_connection_factory_->CreateAudioSource(&constraints);
+    // TODO(perkj): Test audio source when it is implemented. Currently audio
+    // always use the default input.
+    std::string label = stream_label + kAudioTrackLabelBase;
+    return peer_connection_factory_->CreateAudioTrack(label, source);
+  }
+
+  rtc::scoped_refptr<webrtc::VideoTrackInterface> CreateLocalVideoTrack(
+      const std::string& stream_label) {
+    // Set max frame rate to 10fps to reduce the risk of the tests to be flaky.
+    FakeConstraints source_constraints = video_constraints_;
+    source_constraints.SetMandatoryMaxFrameRate(10);
+
+    cricket::FakeVideoCapturer* fake_capturer =
+        new webrtc::FakePeriodicVideoCapturer();
+    video_capturers_.push_back(fake_capturer);
+    rtc::scoped_refptr<webrtc::VideoSourceInterface> source =
+        peer_connection_factory_->CreateVideoSource(fake_capturer,
+                                                    &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(); }
 
   void StopVideoCapturers() {
     for (std::vector<cricket::VideoCapturer*>::iterator it =
              video_capturers_.begin();
@@ skipping 294 matching lines @@
       return false;
     }
     if (options) {
       peer_connection_factory_->SetOptions(*options);
     }
     peer_connection_ = CreatePeerConnection(allocator_factory_.get(),
                                             constraints);
     return peer_connection_.get() != nullptr;
   }
 
-  rtc::scoped_refptr<webrtc::VideoTrackInterface>
-  CreateLocalVideoTrack(const std::string stream_label) {
-    // Set max frame rate to 10fps to reduce the risk of the tests to be flaky.
-    FakeConstraints source_constraints = video_constraints_;
-    source_constraints.SetMandatoryMaxFrameRate(10);
-
-    cricket::FakeVideoCapturer* fake_capturer =
-        new webrtc::FakePeriodicVideoCapturer();
-    video_capturers_.push_back(fake_capturer);
-    rtc::scoped_refptr<webrtc::VideoSourceInterface> source =
-        peer_connection_factory_->CreateVideoSource(
-            fake_capturer, &source_constraints);
-    std::string label = stream_label + kVideoTrackLabelBase;
-    return peer_connection_factory_->CreateVideoTrack(label, source);
-  }
-
   rtc::scoped_refptr<webrtc::PeerConnectionInterface> CreatePeerConnection(
       webrtc::PortAllocatorFactoryInterface* factory,
       const MediaConstraintsInterface* constraints) {
     // CreatePeerConnection with IceServers.
     webrtc::PeerConnectionInterface::IceServers ice_servers;
     webrtc::PeerConnectionInterface::IceServer ice_server;
     ice_server.uri = "stun:stun.l.google.com:19302";
     ice_servers.push_back(ice_server);
 
     rtc::scoped_ptr<webrtc::DtlsIdentityStoreInterface> dtls_identity_store(
@@ skipping 250 matching lines @@
   }
 
   // This test sets up a call between two parties. Both parties send static
   // frames to each other. Once the test is finished the number of sent frames
   // is compared to the number of received frames.
   void LocalP2PTest() {
     if (initiating_client_->NumberOfLocalMediaStreams() == 0) {
       initiating_client_->AddMediaStream(true, true);
     }
     initiating_client_->Negotiate();
-    const int kMaxWaitForActivationMs = 5000;
     // 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;
     // TODO(ronghuawu): Add test to cover the case of sendonly and recvonly.
     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;
@@ skipping 589 matching lines @@
 // VideoDecoderFactory.
 // TODO(holmer): Disabled due to sometimes crashing on buildbots.
 // See issue webrtc/2378.
 TEST_F(JsepPeerConnectionP2PTestClient,
        DISABLED_LocalP2PTestWithVideoDecoderFactory) {
   ASSERT_TRUE(CreateTestClients());
   EnableVideoDecoderFactory();
   LocalP2PTest();
 }
 
+// This tests that if we negotiate after calling CreateSender but before we
+// have a track, then set a track later, frames from the newly-set track are
+// received end-to-end.
+TEST_F(JsepPeerConnectionP2PTestClient, EarlyWarmupTest) {
+  ASSERT_TRUE(CreateTestClients());
+  auto audio_sender = initializing_client()->pc()->CreateSender("audio");
+  auto video_sender = initializing_client()->pc()->CreateSender("video");
+  initializing_client()->Negotiate();
+  // Wait for ICE connection to complete, without any tracks.
+  // Note that the receiving client WILL (in HandleIncomingOffer) create
+  // tracks, so it's only the initiator here that's doing early warmup.
+  ASSERT_TRUE_WAIT(SessionActive(), kMaxWaitForActivationMs);
+  VerifySessionDescriptions();
+  EXPECT_EQ_WAIT(webrtc::PeerConnectionInterface::kIceConnectionCompleted,
+                 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),
+                   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());
   }
 
   bool ParseUrl(const std::string& url,
                 const std::string& username,
                 const std::string& password) {
@@ skipping 157 matching lines @@
   server.urls.push_back("stun:hostname");
   server.urls.push_back("turn:hostname");
   servers.push_back(server);
   EXPECT_TRUE(webrtc::ParseIceServers(servers, &stun_configurations_,
                                       &turn_configurations_));
   EXPECT_EQ(1U, stun_configurations_.size());
   EXPECT_EQ(1U, turn_configurations_.size());
 }
 
 #endif // if !defined(THREAD_SANITIZER)