Reverse |rtx_payload_types| map, and rename.

webrtc/video/video_quality_test.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.
  */
 #include "webrtc/video/video_quality_test.h"
@@ skipping 1462 matching lines @@
   video_encoder_config_.spatial_layers = params_.ss.spatial_layers;
 
   CreateMatchingReceiveConfigs(recv_transport);
 
   const bool decode_all_receive_streams =
       params_.ss.selected_stream == params_.ss.streams.size();
 
   for (size_t i = 0; i < num_video_streams; ++i) {
     video_receive_configs_[i].rtp.nack.rtp_history_ms = kNackRtpHistoryMs;
     video_receive_configs_[i].rtp.rtx_ssrc = kSendRtxSsrcs[i];
-    video_receive_configs_[i].rtp.rtx_payload_types[payload_type] =
-        kSendRtxPayloadType;
+    video_receive_configs_[i].rtp.media_pt_by_rtx_pt[kSendRtxPayloadType] =
+        payload_type;
     video_receive_configs_[i].rtp.transport_cc = params_.call.send_side_bwe;
     video_receive_configs_[i].rtp.remb = !params_.call.send_side_bwe;
     // Enable RTT calculation so NTP time estimator will work.
     video_receive_configs_[i].rtp.rtcp_xr.receiver_reference_time_report = true;
     // Force fake decoders on non-selected simulcast streams.
     if (!decode_all_receive_streams && i != params_.ss.selected_stream) {
       VideoReceiveStream::Decoder decoder;
       decoder.decoder = new test::FakeDecoder();
       decoder.payload_type = video_send_config_.encoder_settings.payload_type;
       decoder.payload_name = video_send_config_.encoder_settings.payload_name;
@@ skipping 121 matching lines @@
         test::CreateMatchingDecoder(thumbnail_send_config.encoder_settings);
     allocated_decoders_.push_back(
         std::unique_ptr<VideoDecoder>(decoder.decoder));
     thumbnail_receive_config.decoders.clear();
     thumbnail_receive_config.decoders.push_back(decoder);
     thumbnail_receive_config.rtp.remote_ssrc =
         thumbnail_send_config.rtp.ssrcs[0];
 
     thumbnail_receive_config.rtp.nack.rtp_history_ms = kNackRtpHistoryMs;
     thumbnail_receive_config.rtp.rtx_ssrc = kThumbnailRtxSsrcStart + i;
-    thumbnail_receive_config.rtp.rtx_payload_types[kPayloadTypeVP8] =
-        kSendRtxPayloadType;
+    thumbnail_receive_config.rtp.media_pt_by_rtx_pt[kSendRtxPayloadType] =
+        kPayloadTypeVP8;
     thumbnail_receive_config.rtp.transport_cc = params_.call.send_side_bwe;
     thumbnail_receive_config.rtp.remb = !params_.call.send_side_bwe;
 
     thumbnail_encoder_configs_.push_back(thumbnail_encoder_config.Copy());
     thumbnail_send_configs_.push_back(thumbnail_send_config.Copy());
     thumbnail_receive_configs_.push_back(thumbnail_receive_config.Copy());
   }
 
   for (int i = 0; i < params_.call.num_thumbnails; ++i) {
     thumbnail_send_streams_.push_back(receiver_call_->CreateVideoSendStream(
@@ skipping 481 matching lines @@
   if (!params_.logging.encoded_frame_base_path.empty()) {
     std::ostringstream str;
     str << receive_logs_++;
     std::string path =
         params_.logging.encoded_frame_base_path + "." + str.str() + ".recv.ivf";
     stream->EnableEncodedFrameRecording(rtc::CreatePlatformFile(path),
                                         100000000);
   }
 }
 }  // namespace webrtc