Add tests for verifying transport feedback for audio and video.

BUG=webrtc:5263
R=mflodman@webrtc.org

Review URL: https://codereview.webrtc.org/1589523002 .

Cr-Commit-Position: refs/heads/master@{#11255}

webrtc/test/call_test.cc

@@ -31,7 +31,7 @@ CallTest::CallTest()
       audio_send_config_(nullptr),
       audio_send_stream_(nullptr),
       fake_encoder_(clock_),
-      num_video_streams_(0),
+      num_video_streams_(1),
       num_audio_streams_(0),
       fake_send_audio_device_(nullptr),
       fake_recv_audio_device_(nullptr) {}
@@ -60,9 +60,9 @@ void CallTest::RunBaseTest(BaseTest* test) {
     }
     CreateReceiverCall(recv_config);
   }
+  test->OnCallsCreated(sender_call_.get(), receiver_call_.get());
   send_transport_.reset(test->CreateSendTransport(sender_call_.get()));
   receive_transport_.reset(test->CreateReceiveTransport());
-  test->OnCallsCreated(sender_call_.get(), receiver_call_.get());
 
   if (test->ShouldCreateReceivers()) {
     send_transport_->SetReceiver(receiver_call_->Receiver());
@@ -97,8 +97,10 @@ void CallTest::RunBaseTest(BaseTest* test) {
     test->OnAudioStreamsCreated(audio_send_stream_, audio_receive_streams_);
   }
 
+  if (num_video_streams_ > 0) {
     CreateFrameGeneratorCapturer();
     test->OnFrameGeneratorCapturerCreated(frame_generator_capturer_.get());
+  }
 
   Start();
   test->PerformTest();
@@ -179,17 +181,20 @@ void CallTest::CreateSendConfig(size_t num_video_streams,
   RTC_DCHECK(num_video_streams <= kNumSsrcs);
   RTC_DCHECK_LE(num_audio_streams, 1u);
   RTC_DCHECK(num_audio_streams == 0 || voe_send_.channel_id >= 0);
+  if (num_video_streams > 0) {
     video_send_config_ = VideoSendStream::Config(send_transport);
     video_send_config_.encoder_settings.encoder = &fake_encoder_;
     video_send_config_.encoder_settings.payload_name = "FAKE";
-  video_send_config_.encoder_settings.payload_type = kFakeVideoSendPayloadType;
+    video_send_config_.encoder_settings.payload_type =
+        kFakeVideoSendPayloadType;
     video_send_config_.rtp.extensions.push_back(
         RtpExtension(RtpExtension::kAbsSendTime, kAbsSendTimeExtensionId));
     video_encoder_config_.streams = test::CreateVideoStreams(num_video_streams);
     for (size_t i = 0; i < num_video_streams; ++i)
       video_send_config_.rtp.ssrcs.push_back(kVideoSendSsrcs[i]);
-  video_send_config_.rtp.extensions.push_back(
+    video_send_config_.rtp.extensions.push_back(RtpExtension(
-      RtpExtension(RtpExtension::kVideoRotation, kVideoRotationRtpExtensionId));
+        RtpExtension::kVideoRotation, kVideoRotationRtpExtensionId));
+  }
 
   if (num_audio_streams > 0) {
     audio_send_config_ = AudioSendStream::Config(send_transport);
@@ -199,10 +204,10 @@ void CallTest::CreateSendConfig(size_t num_video_streams,
 }
 
 void CallTest::CreateMatchingReceiveConfigs(Transport* rtcp_send_transport) {
-  RTC_DCHECK(!video_send_config_.rtp.ssrcs.empty());
   RTC_DCHECK(video_receive_configs_.empty());
   RTC_DCHECK(allocated_decoders_.empty());
-  RTC_DCHECK(num_audio_streams_ == 0 || voe_send_.channel_id >= 0);
+  if (num_video_streams_ > 0) {
+    RTC_DCHECK(!video_send_config_.rtp.ssrcs.empty());
     VideoReceiveStream::Config video_config(rtcp_send_transport);
     video_config.rtp.remb = true;
     video_config.rtp.local_ssrc = kReceiverLocalVideoSsrc;
@@ -217,9 +222,11 @@ void CallTest::CreateMatchingReceiveConfigs(Transport* rtcp_send_transport) {
       video_config.rtp.remote_ssrc = video_send_config_.rtp.ssrcs[i];
       video_receive_configs_.push_back(video_config);
     }
+  }
 
   RTC_DCHECK(num_audio_streams_ <= 1);
   if (num_audio_streams_ == 1) {
+    RTC_DCHECK(voe_send_.channel_id >= 0);
     AudioReceiveStream::Config audio_config;
     audio_config.rtp.local_ssrc = kReceiverLocalAudioSsrc;
     audio_config.rtcp_send_transport = rtcp_send_transport;

webrtc/video/end_to_end_tests.cc

@@ -1542,82 +1542,115 @@ TEST_F(EndToEndTest, AssignsTransportSequenceNumbers) {
   tester.RunTest();
 }
 
-void TransportFeedbackTest(bool feedback_enabled) {
+class TransportFeedbackTester : public test::EndToEndTest {
-  static const int kExtensionId = 5;
-  class TransportFeedbackObserver : public test::DirectTransport {
  public:
-    TransportFeedbackObserver(Call* receiver_call, rtc::Event* done_event)
+  explicit TransportFeedbackTester(bool feedback_enabled,
-        : DirectTransport(receiver_call), done_(done_event) {}
+                                   size_t num_video_streams,
-    virtual ~TransportFeedbackObserver() {}
+                                   size_t num_audio_streams)
+      : EndToEndTest(::webrtc::EndToEndTest::kDefaultTimeoutMs),
+        feedback_enabled_(feedback_enabled),
+        num_video_streams_(num_video_streams),
+        num_audio_streams_(num_audio_streams) {
+    // Only one stream of each supported for now.
+    EXPECT_LE(num_video_streams, 1u);
+    EXPECT_LE(num_audio_streams, 1u);
+  }
 
-    bool SendRtcp(const uint8_t* data, size_t length) override {
+ protected:
+  Action OnSendRtcp(const uint8_t* data, size_t length) override {
+    EXPECT_FALSE(HasTransportFeedback(data, length));
+    return SEND_PACKET;
+  }
+
+  Action OnReceiveRtcp(const uint8_t* data, size_t length) override {
+    if (HasTransportFeedback(data, length))
+      observation_complete_.Set();
+    return SEND_PACKET;
+  }
+
+  bool HasTransportFeedback(const uint8_t* data, size_t length) const {
     RTCPUtility::RTCPParserV2 parser(data, length, true);
     EXPECT_TRUE(parser.IsValid());
 
     RTCPUtility::RTCPPacketTypes packet_type = parser.Begin();
     while (packet_type != RTCPUtility::RTCPPacketTypes::kInvalid) {
-        if (packet_type == RTCPUtility::RTCPPacketTypes::kTransportFeedback) {
+      if (packet_type == RTCPUtility::RTCPPacketTypes::kTransportFeedback)
-          done_->Set();
+        return true;
-          break;
-        }
       packet_type = parser.Iterate();
     }
 
-      return test::DirectTransport::SendRtcp(data, length);
+    return false;
   }
 
-    rtc::Event* done_;
+  void PerformTest() override {
-  };
-
-  class TransportFeedbackTester : public MultiStreamTest {
-   public:
-    explicit TransportFeedbackTester(bool feedback_enabled)
-        : feedback_enabled_(feedback_enabled), done_(false, false) {}
-    virtual ~TransportFeedbackTester() {}
-
-   protected:
-    void Wait() override {
     const int64_t kDisabledFeedbackTimeoutMs = 5000;
-      EXPECT_EQ(feedback_enabled_, done_.Wait(feedback_enabled_
+    EXPECT_EQ(feedback_enabled_,
+              observation_complete_.Wait(feedback_enabled_
                                              ? test::CallTest::kDefaultTimeoutMs
                                              : kDisabledFeedbackTimeoutMs));
   }
 
-    void UpdateSendConfig(
+  void OnCallsCreated(Call* sender_call, Call* receiver_call) override {
-        size_t stream_index,
+    receiver_call_ = receiver_call;
+  }
+
+  size_t GetNumVideoStreams() const override { return num_video_streams_; }
+  size_t GetNumAudioStreams() const override { return num_audio_streams_; }
+
+  void ModifyVideoConfigs(
       VideoSendStream::Config* send_config,
-        VideoEncoderConfig* encoder_config,
+      std::vector<VideoReceiveStream::Config>* receive_configs,
-        test::FrameGeneratorCapturer** frame_generator) override {
+      VideoEncoderConfig* encoder_config) override {
+    send_config->rtp.extensions.clear();
     send_config->rtp.extensions.push_back(
         RtpExtension(RtpExtension::kTransportSequenceNumber, kExtensionId));
+    (*receive_configs)[0].rtp.extensions = send_config->rtp.extensions;
+    (*receive_configs)[0].rtp.transport_cc = feedback_enabled_;
   }
 
-    void UpdateReceiveConfig(
+  void ModifyAudioConfigs(
-        size_t stream_index,
+      AudioSendStream::Config* send_config,
-        VideoReceiveStream::Config* receive_config) override {
+      std::vector<AudioReceiveStream::Config>* receive_configs) override {
-      receive_config->rtp.extensions.push_back(
+    send_config->rtp.extensions.clear();
+    send_config->rtp.extensions.push_back(
         RtpExtension(RtpExtension::kTransportSequenceNumber, kExtensionId));
-      receive_config->rtp.transport_cc = feedback_enabled_;
+    (*receive_configs)[0].rtp.extensions.clear();
-    }
+    (*receive_configs)[0].rtp.extensions = send_config->rtp.extensions;
-
+    (*receive_configs)[0].rtp.transport_cc = feedback_enabled_;
-    test::DirectTransport* CreateReceiveTransport(
+    (*receive_configs)[0].combined_audio_video_bwe = true;
-        Call* receiver_call) override {
-      return new TransportFeedbackObserver(receiver_call, &done_);
   }
 
  private:
+  static const int kExtensionId = 5;
   const bool feedback_enabled_;
-    rtc::Event done_;
+  const size_t num_video_streams_;
-  } tester(feedback_enabled);
+  const size_t num_audio_streams_;
-  tester.RunTest();
+  Call* receiver_call_;
+};
+
+TEST_F(EndToEndTest, VideoReceivesTransportFeedback) {
+  TransportFeedbackTester test(true, 1, 0);
+  RunBaseTest(&test);
 }
 
-TEST_F(EndToEndTest, ReceivesTransportFeedback) {
+TEST_F(EndToEndTest, VideoTransportFeedbackNotConfigured) {
-  TransportFeedbackTest(true);
+  TransportFeedbackTester test(false, 1, 0);
+  RunBaseTest(&test);
 }
 
-TEST_F(EndToEndTest, TransportFeedbackNotConfigured) {
+TEST_F(EndToEndTest, AudioReceivesTransportFeedback) {
-  TransportFeedbackTest(false);
+  TransportFeedbackTester test(true, 0, 1);
+  RunBaseTest(&test);
+}
+
+TEST_F(EndToEndTest, AudioTransportFeedbackNotConfigured) {
+  TransportFeedbackTester test(false, 0, 1);
+  RunBaseTest(&test);
+}
+
+TEST_F(EndToEndTest, AudioVideoReceivesTransportFeedback) {
+  TransportFeedbackTester test(true, 1, 1);
+  RunBaseTest(&test);
 }
 
 TEST_F(EndToEndTest, ObserversEncodedFrames) {