Unify newapi::RtcpMode and RTCPMethod.

webrtc/call/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 111 matching lines @@
   ASSERT_EQ(rtclog::Event::VIDEO_RECEIVER_CONFIG_EVENT, event.type());
   const rtclog::VideoReceiveConfig& receiver_config =
       event.video_receiver_config();
   // Check SSRCs.
   ASSERT_TRUE(receiver_config.has_remote_ssrc());
   EXPECT_EQ(config.rtp.remote_ssrc, receiver_config.remote_ssrc());
   ASSERT_TRUE(receiver_config.has_local_ssrc());
   EXPECT_EQ(config.rtp.local_ssrc, receiver_config.local_ssrc());
   // Check RTCP settings.
   ASSERT_TRUE(receiver_config.has_rtcp_mode());
-  if (config.rtp.rtcp_mode == newapi::kRtcpCompound)
+  if (config.rtp.rtcp_mode == RtcpMode::kCompound)
     EXPECT_EQ(rtclog::VideoReceiveConfig::RTCP_COMPOUND,
               receiver_config.rtcp_mode());
   else
     EXPECT_EQ(rtclog::VideoReceiveConfig::RTCP_REDUCEDSIZE,
               receiver_config.rtcp_mode());
   ASSERT_TRUE(receiver_config.has_receiver_reference_time_report());
   EXPECT_EQ(config.rtp.rtcp_xr.receiver_reference_time_report,
             receiver_config.receiver_reference_time_report());
   ASSERT_TRUE(receiver_config.has_remb());
   EXPECT_EQ(config.rtp.remb, receiver_config.remb());
@@ skipping 206 matching lines @@
                                 VideoReceiveStream::Config* config) {
   // Create a map from a payload type to an encoder name.
   VideoReceiveStream::Decoder decoder;
   decoder.payload_type = rand();
   decoder.payload_name = (rand() % 2 ? "VP8" : "H264");
   config->decoders.push_back(decoder);
   // Add SSRCs for the stream.
   config->rtp.remote_ssrc = rand();
   config->rtp.local_ssrc = rand();
   // Add extensions and settings for RTCP.
-  config->rtp.rtcp_mode = rand() % 2 ? newapi::kRtcpCompound
-                                     : newapi::kRtcpReducedSize;
+  config->rtp.rtcp_mode =
+      rand() % 2 ? RtcpMode::kCompound : RtcpMode::kReducedSize;
   config->rtp.rtcp_xr.receiver_reference_time_report = (rand() % 2 == 1);
   config->rtp.remb = (rand() % 2 == 1);
   // 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 = rand();
   rtx_pair.payload_type = rand();
   config->rtp.rtx.insert(std::make_pair(rand(), rtx_pair));
   // Add header extensions.
   for (unsigned i = 0; i < kNumExtensions; i++) {
     if (extensions_bitvector & (1u << i)) {
@@ skipping 174 matching lines @@
                             extensions,       // Bit vector choosing extensions
                             csrcs_count,      // Number of contributing sources
                             rand());
     }
   }
 }
 
 }  // namespace webrtc
 
 #endif  // ENABLE_RTC_EVENT_LOG