Require negotiation to send transport cc feedback over RTCP.

webrtc/video/rampup_tests.cc

 /*
  *  Copyright (c) 2013 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 106 matching lines @@
     // the last one.
     expected_bitrate_bps_ = encoder_config->streams.back().min_bitrate_bps;
     for (size_t i = 0; i < encoder_config->streams.size() - 1; ++i) {
       expected_bitrate_bps_ += encoder_config->streams[i].target_bitrate_bps;
     }
   }
 
   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));
   }
 
   send_config->rtp.nack.rtp_history_ms = test::CallTest::kNackRtpHistoryMs;
   send_config->rtp.ssrcs = ssrcs_;
   if (rtx_) {
     send_config->rtp.rtx.payload_type = test::CallTest::kSendRtxPayloadType;
     send_config->rtp.rtx.ssrcs = rtx_ssrcs_;
   }
   if (red_) {
     send_config->rtp.fec.ulpfec_payload_type =
         test::CallTest::kUlpfecPayloadType;
     send_config->rtp.fec.red_payload_type = test::CallTest::kRedPayloadType;
   }
 
   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];
     recv_config.rtp.nack.rtp_history_ms = send_config->rtp.nack.rtp_history_ms;
 
     if (red_) {
       recv_config.rtp.fec.red_payload_type =
           send_config->rtp.fec.red_payload_type;
       recv_config.rtp.fec.ulpfec_payload_type =
           send_config->rtp.fec.ulpfec_payload_type;
@@ skipping 332 matching lines @@
   RampUpTester test(3, 0, RtpExtension::kTransportSequenceNumber, true, true);
   RunBaseTest(&test, FakeNetworkPipe::Config());
 }
 
 TEST_F(RampUpTest, TransportSequenceNumberSingleStreamWithHighStartBitrate) {
   RampUpTester test(1, 0.9 * kSingleStreamTargetBps,
                     RtpExtension::kTransportSequenceNumber, false, false);
   RunBaseTest(&test, FakeNetworkPipe::Config());
 }
 }  // namespace webrtc