| Index: webrtc/audio/audio_send_stream_unittest.cc
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| diff --git a/webrtc/audio/audio_send_stream_unittest.cc b/webrtc/audio/audio_send_stream_unittest.cc
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| index 917206470530031a2c397b4c0f2a87699b1cf274..ebac828079b0e6c42e15517666de5e4777cee043 100644
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| --- a/webrtc/audio/audio_send_stream_unittest.cc
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| +++ b/webrtc/audio/audio_send_stream_unittest.cc
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| @@ -106,6 +106,10 @@ struct ConfigHelper {
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|                .Times(1);
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|            EXPECT_CALL(*channel_proxy_, DeRegisterExternalTransport())
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|                .Times(1);
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| +          EXPECT_CALL(*channel_proxy_, SetRtcEventLog(testing::NotNull()))
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| +              .Times(1);
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| +          EXPECT_CALL(*channel_proxy_, SetRtcEventLog(testing::IsNull()))
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| +              .Times(1);  // Destructor resets the event log
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|            return channel_proxy_;
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|          }));
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|      stream_config_.voe_channel_id = kChannelId;
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| @@ -128,6 +132,7 @@ struct ConfigHelper {
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|    }
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|    BitrateAllocator* bitrate_allocator() { return &bitrate_allocator_; }
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|    rtc::TaskQueue* worker_queue() { return &worker_queue_; }
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| +  RtcEventLog* event_log() { return &event_log_; }
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|  
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|    void SetupMockForSendTelephoneEvent() {
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|      EXPECT_TRUE(channel_proxy_);
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| @@ -210,14 +215,16 @@ TEST(AudioSendStreamTest, ConstructDestruct) {
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|    ConfigHelper helper;
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|    internal::AudioSendStream send_stream(
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|        helper.config(), helper.audio_state(), helper.worker_queue(),
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| -      helper.congestion_controller(), helper.bitrate_allocator());
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| +      helper.congestion_controller(), helper.bitrate_allocator(),
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| +      helper.event_log());
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|  }
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|  
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|  TEST(AudioSendStreamTest, SendTelephoneEvent) {
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|    ConfigHelper helper;
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|    internal::AudioSendStream send_stream(
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|        helper.config(), helper.audio_state(), helper.worker_queue(),
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| -      helper.congestion_controller(), helper.bitrate_allocator());
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| +      helper.congestion_controller(), helper.bitrate_allocator(),
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| +      helper.event_log());
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|    helper.SetupMockForSendTelephoneEvent();
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|    EXPECT_TRUE(send_stream.SendTelephoneEvent(kTelephoneEventPayloadType,
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|        kTelephoneEventCode, kTelephoneEventDuration));
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| @@ -227,7 +234,8 @@ TEST(AudioSendStreamTest, SetMuted) {
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|    ConfigHelper helper;
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|    internal::AudioSendStream send_stream(
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|        helper.config(), helper.audio_state(), helper.worker_queue(),
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| -      helper.congestion_controller(), helper.bitrate_allocator());
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| +      helper.congestion_controller(), helper.bitrate_allocator(),
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| +      helper.event_log());
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|    EXPECT_CALL(*helper.channel_proxy(), SetInputMute(true));
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|    send_stream.SetMuted(true);
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|  }
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| @@ -236,7 +244,8 @@ TEST(AudioSendStreamTest, GetStats) {
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|    ConfigHelper helper;
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|    internal::AudioSendStream send_stream(
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|        helper.config(), helper.audio_state(), helper.worker_queue(),
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| -      helper.congestion_controller(), helper.bitrate_allocator());
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| +      helper.congestion_controller(), helper.bitrate_allocator(),
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| +      helper.event_log());
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|    helper.SetupMockForGetStats();
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|    AudioSendStream::Stats stats = send_stream.GetStats();
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|    EXPECT_EQ(kSsrc, stats.local_ssrc);
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| @@ -265,7 +274,8 @@ TEST(AudioSendStreamTest, GetStatsTypingNoiseDetected) {
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|    ConfigHelper helper;
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|    internal::AudioSendStream send_stream(
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|        helper.config(), helper.audio_state(), helper.worker_queue(),
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| -      helper.congestion_controller(), helper.bitrate_allocator());
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| +      helper.congestion_controller(), helper.bitrate_allocator(),
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| +      helper.event_log());
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|    helper.SetupMockForGetStats();
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|    EXPECT_FALSE(send_stream.GetStats().typing_noise_detected);
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|  
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| 
 |