Reverse |rtx_payload_types| map, and rename.

webrtc/call/call.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 68 matching lines @@
 }
 
 bool UseSendSideBwe(const AudioReceiveStream::Config& config) {
   return UseSendSideBwe(config.rtp.extensions, config.rtp.transport_cc);
 }
 
 bool UseSendSideBwe(const FlexfecReceiveStream::Config& config) {
   return UseSendSideBwe(config.rtp_header_extensions, config.transport_cc);
 }
 
+const int* FindKeyByValue(const std::map<int, int>& m, int v) {
+  for (const auto& kv : m) {
+    if (kv.second == v)
+      return &kv.first;
+  }
+  return nullptr;
+}
+
 rtclog::StreamConfig CreateRtcLogStreamConfig(
     const VideoReceiveStream::Config& config) {
   rtclog::StreamConfig rtclog_config;
   rtclog_config.remote_ssrc = config.rtp.remote_ssrc;
   rtclog_config.local_ssrc = config.rtp.local_ssrc;
   rtclog_config.rtx_ssrc = config.rtp.rtx_ssrc;
   rtclog_config.rtcp_mode = config.rtp.rtcp_mode;
   rtclog_config.remb = config.rtp.remb;
   rtclog_config.rtp_extensions = config.rtp.extensions;
 
   for (const auto& d : config.decoders) {
-    auto search = config.rtp.rtx_payload_types.find(d.payload_type);
-    rtclog_config.codecs.emplace_back(
-        d.payload_name, d.payload_type,
-        search != config.rtp.rtx_payload_types.end() ? search->second : 0);
+    const int* search =
+        FindKeyByValue(config.rtp.rtx_associated_payload_types, d.payload_type);
+    rtclog_config.codecs.emplace_back(d.payload_name, d.payload_type,
+                                      search ? *search : 0);
   }
   return rtclog_config;
 }
 
 rtclog::StreamConfig CreateRtcLogStreamConfig(
     const VideoSendStream::Config& config,
     size_t ssrc_index) {
   rtclog::StreamConfig rtclog_config;
   rtclog_config.local_ssrc = config.rtp.ssrcs[ssrc_index];
   if (ssrc_index < config.rtp.rtx.ssrcs.size()) {
@@ skipping 1319 matching lines @@
       (use_send_side_bwe && header.extension.hasTransportSequenceNumber)) {
     receive_side_cc_.OnReceivedPacket(
         packet.arrival_time_ms(), packet.payload_size() + packet.padding_size(),
         header);
   }
 }
 
 }  // namespace internal
 
 }  // namespace webrtc