Move RtcEventLog object from inside VoiceEngine to Call.

webrtc/audio/audio_receive_stream_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 48 matching lines @@
     123, "codec_name_recv", 96000, -187, 0, -103};
 const NetworkStatistics kNetworkStats = {
     123, 456, false, 0, 0, 789, 12, 345, 678, 901, -1, -1, -1, -1, -1, 0};
 const AudioDecodingCallStats kAudioDecodeStats = MakeAudioDecodeStatsForTest();
 
 struct ConfigHelper {
   ConfigHelper()
       : simulated_clock_(123456),
         congestion_controller_(&simulated_clock_,
                                &bitrate_observer_,
-                               &remote_bitrate_observer_) {
+                               &remote_bitrate_observer_,
+                               nullptr) {
     using testing::Invoke;
 
     EXPECT_CALL(voice_engine_,
         RegisterVoiceEngineObserver(_)).WillOnce(Return(0));
     EXPECT_CALL(voice_engine_,
         DeRegisterVoiceEngineObserver()).WillOnce(Return(0));
     AudioState::Config config;
     config.voice_engine = &voice_engine_;
     audio_state_ = AudioState::Create(config);
 
@@ skipping 133 matching lines @@
       "{rtp: {remote_ssrc: 1234, local_ssrc: 5678, extensions: [{name: "
       "http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time, id: 2}], "
       "transport_cc: off}, "
       "receive_transport: nullptr, rtcp_send_transport: nullptr, "
       "voe_channel_id: 2}",
       config.ToString());
 }
 
 TEST(AudioReceiveStreamTest, ConstructDestruct) {
   ConfigHelper helper;
-  internal::AudioReceiveStream recv_stream(
-      helper.congestion_controller(), helper.config(), helper.audio_state());
+  internal::AudioReceiveStream recv_stream(helper.congestion_controller(),
+                                           helper.config(),
+                                           helper.audio_state(), nullptr);
 }
 
 MATCHER_P(VerifyHeaderExtension, expected_extension, "") {
   return arg.extension.hasAbsoluteSendTime ==
              expected_extension.hasAbsoluteSendTime &&
          arg.extension.absoluteSendTime ==
              expected_extension.absoluteSendTime &&
          arg.extension.hasTransportSequenceNumber ==
              expected_extension.hasTransportSequenceNumber &&
          arg.extension.transportSequenceNumber ==
              expected_extension.transportSequenceNumber;
 }
 
 TEST(AudioReceiveStreamTest, AudioPacketUpdatesBweFeedback) {
   ConfigHelper helper;
   helper.config().rtp.transport_cc = true;
   helper.SetupMockForBweFeedback(true);
-  internal::AudioReceiveStream recv_stream(
-      helper.congestion_controller(), helper.config(), helper.audio_state());
+  internal::AudioReceiveStream recv_stream(helper.congestion_controller(),
+                                           helper.config(),
+                                           helper.audio_state(), nullptr);
   const int kTransportSequenceNumberValue = 1234;
   std::vector<uint8_t> rtp_packet = CreateRtpHeaderWithOneByteExtension(
       kTransportSequenceNumberId, kTransportSequenceNumberValue, 2);
   PacketTime packet_time(5678000, 0);
   const size_t kExpectedHeaderLength = 20;
   RTPHeaderExtension expected_extension;
   expected_extension.hasTransportSequenceNumber = true;
   expected_extension.transportSequenceNumber = kTransportSequenceNumberValue;
   EXPECT_CALL(*helper.remote_bitrate_estimator(),
               IncomingPacket(packet_time.timestamp / 1000,
                              rtp_packet.size() - kExpectedHeaderLength,
                              VerifyHeaderExtension(expected_extension), false))
       .Times(1);
   EXPECT_TRUE(
       recv_stream.DeliverRtp(&rtp_packet[0], rtp_packet.size(), packet_time));
 }
 
 TEST(AudioReceiveStreamTest, GetStats) {
   ConfigHelper helper;
-  internal::AudioReceiveStream recv_stream(
-      helper.congestion_controller(), helper.config(), helper.audio_state());
+  internal::AudioReceiveStream recv_stream(helper.congestion_controller(),
+                                           helper.config(),
+                                           helper.audio_state(), nullptr);
   helper.SetupMockForGetStats();
   AudioReceiveStream::Stats stats = recv_stream.GetStats();
   EXPECT_EQ(kRemoteSsrc, stats.remote_ssrc);
   EXPECT_EQ(static_cast<int64_t>(kCallStats.bytesReceived), stats.bytes_rcvd);
   EXPECT_EQ(static_cast<uint32_t>(kCallStats.packetsReceived),
             stats.packets_rcvd);
   EXPECT_EQ(kCallStats.cumulativeLost, stats.packets_lost);
   EXPECT_EQ(Q8ToFloat(kCallStats.fractionLost), stats.fraction_lost);
   EXPECT_EQ(std::string(kCodecInst.plname), stats.codec_name);
   EXPECT_EQ(kCallStats.extendedMax, stats.ext_seqnum);
@@ skipping 19 matching lines @@
   EXPECT_EQ(kAudioDecodeStats.calls_to_neteq, stats.decoding_calls_to_neteq);
   EXPECT_EQ(kAudioDecodeStats.decoded_normal, stats.decoding_normal);
   EXPECT_EQ(kAudioDecodeStats.decoded_plc, stats.decoding_plc);
   EXPECT_EQ(kAudioDecodeStats.decoded_cng, stats.decoding_cng);
   EXPECT_EQ(kAudioDecodeStats.decoded_plc_cng, stats.decoding_plc_cng);
   EXPECT_EQ(kCallStats.capture_start_ntp_time_ms_,
             stats.capture_start_ntp_time_ms);
 }
 }  // namespace test
 }  // namespace webrtc