Reland of Adding the ability to create an RtpSender without a track.

talk/app/webrtc/peerconnection_unittest.cc

Index: talk/app/webrtc/peerconnection_unittest.cc
diff --git a/talk/app/webrtc/peerconnection_unittest.cc b/talk/app/webrtc/peerconnection_unittest.cc
index 6307bfc6d88d45819d8ad731d2e1b76a0aeb5625..5aba2460c5e7125cecdd5e5cc5d1e70a452cce3a 100644
--- a/talk/app/webrtc/peerconnection_unittest.cc
+++ b/talk/app/webrtc/peerconnection_unittest.cc
@@ -98,6 +98,7 @@ static const int kMaxWaitMs = 10000;
 #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;
@@ -288,18 +289,7 @@ class PeerConnectionTestClient : public webrtc::PeerConnectionObserver,
         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));
@@ -401,6 +391,35 @@ class PeerConnectionTestClient : public webrtc::PeerConnectionObserver,
     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();
@@ -715,22 +734,6 @@ class PeerConnectionTestClient : public webrtc::PeerConnectionObserver,
     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) {
@@ -1011,7 +1014,6 @@ class MAYBE_JsepPeerConnectionP2PTestClient : public testing::Test {
       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);
@@ -1653,6 +1655,34 @@ TEST_F(MAYBE_JsepPeerConnectionP2PTestClient,
   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(MAYBE_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.