Use RtxReceiveStream.

webrtc/call/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 189 matching lines @@
 
     recv_config.rtp.remote_ssrc = video_ssrcs_[i];
     recv_config.rtp.nack.rtp_history_ms = send_config->rtp.nack.rtp_history_ms;
 
     if (red_) {
       recv_config.rtp.ulpfec.red_payload_type =
           send_config->rtp.ulpfec.red_payload_type;
       recv_config.rtp.ulpfec.ulpfec_payload_type =
           send_config->rtp.ulpfec.ulpfec_payload_type;
       if (rtx_) {
-        recv_config.rtp.ulpfec.red_rtx_payload_type =
-            send_config->rtp.ulpfec.red_rtx_payload_type;
+        recv_config.rtp.rtx_associated_payload_types
+            [send_config->rtp.ulpfec.red_rtx_payload_type] =
+            send_config->rtp.ulpfec.red_payload_type;
       }
     }
 
     if (rtx_) {
       recv_config.rtp.rtx_ssrc = video_rtx_ssrcs_[i];
       recv_config.rtp
           .rtx_associated_payload_types[send_config->rtp.rtx.payload_type] =
           send_config->encoder_settings.payload_type;
     }
     ++i;
@@ skipping 421 matching lines @@
   RunBaseTest(&test);
 }
 
 TEST_F(RampUpTest, AudioTransportSequenceNumber) {
   RampUpTester test(0, 1, 0, 300000, 10000,
                     RtpExtension::kTransportSequenceNumberUri, false, false,
                     false);
   RunBaseTest(&test);
 }
 }  // namespace webrtc