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.
  */
 
 #include <string>
 #include <vector>
 
 #include "testing/gtest/include/gtest/gtest.h"
 
 #include "webrtc/audio/audio_receive_stream.h"
 #include "webrtc/audio/conversion.h"
+#include "webrtc/call/mock/mock_rtc_event_log.h"
 #include "webrtc/modules/bitrate_controller/include/mock/mock_bitrate_controller.h"
 #include "webrtc/modules/congestion_controller/include/mock/mock_congestion_controller.h"
 #include "webrtc/modules/pacing/packet_router.h"
 #include "webrtc/modules/remote_bitrate_estimator/include/mock/mock_remote_bitrate_estimator.h"
 #include "webrtc/modules/rtp_rtcp/source/byte_io.h"
 #include "webrtc/system_wrappers/include/clock.h"
 #include "webrtc/test/mock_voe_channel_proxy.h"
 #include "webrtc/test/mock_voice_engine.h"
 
 namespace webrtc {
@@ skipping 31 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_,
+                               &event_log_) {
     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 11 matching lines @@
           EXPECT_CALL(*channel_proxy_, EnableReceiveTransportSequenceNumber(
                                            kTransportSequenceNumberId))
               .Times(1);
           EXPECT_CALL(*channel_proxy_,
                       RegisterReceiverCongestionControlObjects(&packet_router_))
               .Times(1);
           EXPECT_CALL(congestion_controller_, packet_router())
               .WillOnce(Return(&packet_router_));
           EXPECT_CALL(*channel_proxy_, ResetCongestionControlObjects())
               .Times(1);
+          testing::Expectation expect_set =
+              EXPECT_CALL(*channel_proxy_, SetRtcEventLog(testing::NotNull()))

[the sun, 2016/03/30 15:20:38]

Could check for argument to be &event_log_ ?

[ivoc, 2016/03/31 09:06:56]

Good idea, done.

+                  .Times(1);
+          EXPECT_CALL(*channel_proxy_, SetRtcEventLog(testing::IsNull()))
+              .Times(1)
+              .After(expect_set);
           return channel_proxy_;
         }));
     stream_config_.voe_channel_id = kChannelId;
     stream_config_.rtp.local_ssrc = kLocalSsrc;
     stream_config_.rtp.remote_ssrc = kRemoteSsrc;
     stream_config_.rtp.extensions.push_back(
         RtpExtension(RtpExtension::kAbsSendTime, kAbsSendTimeId));
     stream_config_.rtp.extensions.push_back(
         RtpExtension(RtpExtension::kAudioLevel, kAudioLevelId));
     stream_config_.rtp.extensions.push_back(RtpExtension(
         RtpExtension::kTransportSequenceNumber, kTransportSequenceNumberId));
   }
 
   MockCongestionController* congestion_controller() {
     return &congestion_controller_;
   }
   MockRemoteBitrateEstimator* remote_bitrate_estimator() {
     return &remote_bitrate_estimator_;
   }
+  MockRtcEventLog* event_log() { return &event_log_; }
   AudioReceiveStream::Config& config() { return stream_config_; }
   rtc::scoped_refptr<AudioState> audio_state() { return audio_state_; }
   MockVoiceEngine& voice_engine() { return voice_engine_; }
 
   void SetupMockForBweFeedback(bool send_side_bwe) {
     EXPECT_CALL(congestion_controller_,
                 GetRemoteBitrateEstimator(send_side_bwe))
         .WillOnce(Return(&remote_bitrate_estimator_));
     EXPECT_CALL(remote_bitrate_estimator_,
                 RemoveStream(stream_config_.rtp.remote_ssrc));
@@ skipping 19 matching lines @@
         .WillOnce(DoAll(SetArgReferee<1>(kCodecInst), Return(0)));
   }
 
  private:
   SimulatedClock simulated_clock_;
   PacketRouter packet_router_;
   testing::NiceMock<MockBitrateObserver> bitrate_observer_;
   testing::NiceMock<MockRemoteBitrateObserver> remote_bitrate_observer_;
   MockCongestionController congestion_controller_;
   MockRemoteBitrateEstimator remote_bitrate_estimator_;
+  MockRtcEventLog event_log_;
   testing::StrictMock<MockVoiceEngine> voice_engine_;
   rtc::scoped_refptr<AudioState> audio_state_;
   AudioReceiveStream::Config stream_config_;
   testing::StrictMock<MockVoEChannelProxy>* channel_proxy_ = nullptr;
 };
 
 void BuildOneByteExtension(std::vector<uint8_t>::iterator it,
                            int id,
                            uint32_t extension_value,
                            size_t value_length) {
@@ skipping 45 matching lines @@
       "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());
+      helper.congestion_controller(), helper.config(), helper.audio_state(),
+      helper.event_log());
 }
 
 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());
+      helper.congestion_controller(), helper.config(), helper.audio_state(),
+      helper.event_log());
   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());
+      helper.congestion_controller(), helper.config(), helper.audio_state(),
+      helper.event_log());
   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