Require negotiation to send transport cc feedback over RTCP.

BUG=4312

Review URL: https://codereview.webrtc.org/1452883002

Cr-Commit-Position: refs/heads/master@{#10735}

5 files changed, 31 insertions, 6 deletions

diff --git a/webrtc/video/end_to_end_tests.cc b/webrtc/video/end_to_end_tests.cc
index e2450d9c7d..e541d370d8 100644
--- a/webrtc/video/end_to_end_tests.cc
+++ b/webrtc/video/end_to_end_tests.cc
@@ -1533,9 +1533,8 @@ TEST_F(EndToEndTest, AssignsTransportSequenceNumbers) {
   tester.RunTest();
 }
 
-TEST_F(EndToEndTest, ReceivesTransportFeedback) {
+void TransportFeedbackTest(bool feedback_enabled) {
   static const int kExtensionId = 5;
-
   class TransportFeedbackObserver : public test::DirectTransport {
    public:
     TransportFeedbackObserver(Call* receiver_call, rtc::Event* done_event)
@@ -1563,12 +1562,16 @@ TEST_F(EndToEndTest, ReceivesTransportFeedback) {
 
   class TransportFeedbackTester : public MultiStreamTest {
    public:
-    TransportFeedbackTester() : done_(false, false) {}
+    explicit TransportFeedbackTester(bool feedback_enabled)
+        : feedback_enabled_(feedback_enabled), done_(false, false) {}
     virtual ~TransportFeedbackTester() {}
 
    protected:
     void Wait() override {
-      EXPECT_TRUE(done_.Wait(CallTest::kDefaultTimeoutMs));
+      const int64_t kDisabledFeedbackTimeoutMs = 5000;
+      EXPECT_EQ(feedback_enabled_, done_.Wait(feedback_enabled_
+                                  ? test::CallTest::kDefaultTimeoutMs
+                                  : kDisabledFeedbackTimeoutMs));
     }
 
     void UpdateSendConfig(
@@ -1585,6 +1588,7 @@ TEST_F(EndToEndTest, ReceivesTransportFeedback) {
         VideoReceiveStream::Config* receive_config) override {
       receive_config->rtp.extensions.push_back(
           RtpExtension(RtpExtension::kTransportSequenceNumber, kExtensionId));
+      receive_config->rtp.transport_cc = feedback_enabled_;
     }
 
     test::DirectTransport* CreateReceiveTransport(
@@ -1593,10 +1597,20 @@ TEST_F(EndToEndTest, ReceivesTransportFeedback) {
     }
 
    private:
+    const bool feedback_enabled_;
     rtc::Event done_;
-  } tester;
+  } tester(feedback_enabled);
   tester.RunTest();
 }
+
+TEST_F(EndToEndTest, ReceivesTransportFeedback) {
+  TransportFeedbackTest(true);
+}
+
+TEST_F(EndToEndTest, TransportFeedbackNotConfigured) {
+  TransportFeedbackTest(false);
+}
+
 TEST_F(EndToEndTest, ObserversEncodedFrames) {
   class EncodedFrameTestObserver : public EncodedFrameObserver {
    public:
diff --git a/webrtc/video/rampup_tests.cc b/webrtc/video/rampup_tests.cc
index 6450e0290d..c9b8e11122 100644
--- a/webrtc/video/rampup_tests.cc
+++ b/webrtc/video/rampup_tests.cc
@@ -124,16 +124,20 @@ void RampUpTester::ModifyConfigs(
   send_config->rtp.extensions.clear();
 
   bool remb;
+  bool transport_cc;
   if (extension_type_ == RtpExtension::kAbsSendTime) {
     remb = true;
+    transport_cc = false;
     send_config->rtp.extensions.push_back(
         RtpExtension(extension_type_.c_str(), kAbsSendTimeExtensionId));
   } else if (extension_type_ == RtpExtension::kTransportSequenceNumber) {
     remb = false;
+    transport_cc = true;
     send_config->rtp.extensions.push_back(RtpExtension(
         extension_type_.c_str(), kTransportSequenceNumberExtensionId));
   } else {
     remb = true;
+    transport_cc = false;
     send_config->rtp.extensions.push_back(RtpExtension(
         extension_type_.c_str(), kTransmissionTimeOffsetExtensionId));
   }
@@ -153,6 +157,7 @@ void RampUpTester::ModifyConfigs(
   size_t i = 0;
   for (VideoReceiveStream::Config& recv_config : *receive_configs) {
     recv_config.rtp.remb = remb;
+    recv_config.rtp.transport_cc = transport_cc;
     recv_config.rtp.extensions = send_config->rtp.extensions;
 
     recv_config.rtp.remote_ssrc = ssrcs_[i];
diff --git a/webrtc/video/video_quality_test.cc b/webrtc/video/video_quality_test.cc
index ebceb5642b..a46fe1bd7c 100644
--- a/webrtc/video/video_quality_test.cc
+++ b/webrtc/video/video_quality_test.cc
@@ -828,6 +828,7 @@ void VideoQualityTest::SetupCommon(Transport* send_transport,
     receive_configs_[i].rtp.rtx[kSendRtxPayloadType].ssrc = kSendRtxSsrcs[i];
     receive_configs_[i].rtp.rtx[kSendRtxPayloadType].payload_type =
         kSendRtxPayloadType;
+    receive_configs_[i].rtp.transport_cc = params_.common.send_side_bwe;
   }
 }
 
diff --git a/webrtc/video/video_receive_stream.cc b/webrtc/video/video_receive_stream.cc
index 287f5355dc..683e89aa97 100644
--- a/webrtc/video/video_receive_stream.cc
+++ b/webrtc/video/video_receive_stream.cc
@@ -81,6 +81,7 @@ std::string VideoReceiveStream::Config::Rtp::ToString() const {
      << (rtcp_xr.receiver_reference_time_report ? "on" : "off");
   ss << '}';
   ss << ", remb: " << (remb ? "on" : "off");
+  ss << ", transport_cc: " << (transport_cc ? "on" : "off");
   ss << ", nack: {rtp_history_ms: " << nack.rtp_history_ms << '}';
   ss << ", fec: " << fec.ToString();
   ss << ", rtx: {";
@@ -153,7 +154,8 @@ VideoReceiveStream::VideoReceiveStream(
       call_stats_(call_stats) {
   LOG(LS_INFO) << "VideoReceiveStream: " << config_.ToString();
 
-  bool send_side_bwe = UseSendSideBwe(config_.rtp.extensions);
+  bool send_side_bwe =
+      config.rtp.transport_cc && UseSendSideBwe(config_.rtp.extensions);
 
   RemoteBitrateEstimator* bitrate_estimator =
       congestion_controller_->GetRemoteBitrateEstimator(send_side_bwe);
diff --git a/webrtc/video_receive_stream.h b/webrtc/video_receive_stream.h
index 275162ca1c..6be2c5a86e 100644
--- a/webrtc/video_receive_stream.h
+++ b/webrtc/video_receive_stream.h
@@ -113,6 +113,9 @@ class VideoReceiveStream : public ReceiveStream {
       // See draft-alvestrand-rmcat-remb for information.
       bool remb = false;
 
+      // See draft-holmer-rmcat-transport-wide-cc-extensions for details.
+      bool transport_cc = false;
+
       // See NackConfig for description.
       NackConfig nack;