Make RTX pt/apt reconfigurable by calling WebRtcVideoChannel2::SetRecvParameters.

webrtc/logging/rtc_event_log/rtc_event_log_unittest.cc

 /*
  *  Copyright (c) 2015 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 147 matching lines @@
   decoder.payload_type = prng->Rand(0, 127);
   decoder.payload_name = (prng->Rand<bool>() ? "VP8" : "H264");
   config->decoders.push_back(decoder);
   // Add SSRCs for the stream.
   config->rtp.remote_ssrc = prng->Rand<uint32_t>();
   config->rtp.local_ssrc = prng->Rand<uint32_t>();
   // Add extensions and settings for RTCP.
   config->rtp.rtcp_mode =
       prng->Rand<bool>() ? RtcpMode::kCompound : RtcpMode::kReducedSize;
   config->rtp.remb = prng->Rand<bool>();
-  // Add a map from a payload type to a new ssrc and a new payload type for RTX.
-  VideoReceiveStream::Config::Rtp::Rtx rtx_pair;
-  rtx_pair.ssrc = prng->Rand<uint32_t>();
-  rtx_pair.payload_type = prng->Rand(0, 127);
-  config->rtp.rtx.insert(std::make_pair(prng->Rand(0, 127), rtx_pair));
+  config->rtp.rtx_ssrc = prng->Rand<uint32_t>();
+  // Add a map from a payload type to a new payload type for RTX.
+  config->rtp.rtx_payload_types.insert(
+      std::make_pair(prng->Rand(0, 127), prng->Rand(0, 127)));
   // Add header extensions.
   for (unsigned i = 0; i < kNumExtensions; i++) {
     if (extensions_bitvector & (1u << i)) {
       config->rtp.extensions.push_back(
           RtpExtension(kExtensionNames[i], prng->Rand<int>()));
     }
   }
 }
 
 void GenerateVideoSendConfig(uint32_t extensions_bitvector,
@@ skipping 469 matching lines @@
   VideoSendConfigReadWriteTest test;
   test.DoTest();
 }
 
 TEST(RtcEventLogTest, LogAudioNetworkAdaptation) {
   AudioNetworkAdaptationReadWriteTest test;
   test.DoTest();
 }
 
 }  // namespace webrtc