Explicitly enable RED over RTX in rampup tests.

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 39 matching lines @@
       test_start_ms_(-1),
       ramp_up_finished_ms_(-1),
       extension_type_(extension_type),
       video_ssrcs_(GenerateSsrcs(num_video_streams_, 100)),
       video_rtx_ssrcs_(GenerateSsrcs(num_video_streams_, 200)),
       audio_ssrcs_(GenerateSsrcs(num_audio_streams_, 300)),
       poller_thread_(&BitrateStatsPollingThread,
                      this,
                      "BitrateStatsPollingThread") {
   EXPECT_LE(num_audio_streams_, 1u);
-  if (rtx_) {
-    for (size_t i = 0; i < video_ssrcs_.size(); ++i)
-      rtx_ssrc_map_[video_rtx_ssrcs_[i]] = video_ssrcs_[i];
-  }
 }
 
 RampUpTester::~RampUpTester() {
   event_.Set();
 }
 
 Call::Config RampUpTester::GetSenderCallConfig() {
   Call::Config call_config(&event_log_);
   if (start_bitrate_bps_ != 0) {
     call_config.bitrate_config.start_bitrate_bps = start_bitrate_bps_;
@@ skipping 91 matching lines @@
   send_config->rtp.nack.rtp_history_ms = test::CallTest::kNackRtpHistoryMs;
   send_config->rtp.ssrcs = video_ssrcs_;
   if (rtx_) {
     send_config->rtp.rtx.payload_type = test::CallTest::kSendRtxPayloadType;
     send_config->rtp.rtx.ssrcs = video_rtx_ssrcs_;
   }
   if (red_) {
     send_config->rtp.ulpfec.ulpfec_payload_type =
         test::CallTest::kUlpfecPayloadType;
     send_config->rtp.ulpfec.red_payload_type = test::CallTest::kRedPayloadType;
+    if (rtx_) {
+      send_config->rtp.ulpfec.red_rtx_payload_type =
+          test::CallTest::kRtxRedPayloadType;
+    }
   }
 
   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 = 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;
+      }
     }
 
     if (rtx_) {
       recv_config.rtp.rtx[send_config->encoder_settings.payload_type].ssrc =
           video_rtx_ssrcs_[i];
       recv_config.rtp.rtx[send_config->encoder_settings.payload_type]
           .payload_type = send_config->rtp.rtx.payload_type;
     }
     ++i;
   }
@@ skipping 419 matching lines @@
                     true);
   RunBaseTest(&test);
 }
 
 TEST_F(RampUpTest, TransportSequenceNumberSingleStreamWithHighStartBitrate) {
   RampUpTester test(1, 0, 0.9 * kSingleStreamTargetBps,
                     RtpExtension::kTransportSequenceNumberUri, false, false);
   RunBaseTest(&test);
 }
 }  // namespace webrtc