Update smoothed bitrate.

webrtc/audio/audio_send_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 199 matching lines @@
     audio_processing_stats_.echo_return_loss_enhancement.Set(
         kEchoReturnLossEnhancement, kEchoReturnLossEnhancement,
         kEchoReturnLossEnhancement, kEchoReturnLossEnhancement);
     audio_processing_stats_.delay_median = kEchoDelayMedian;
     audio_processing_stats_.delay_standard_deviation = kEchoDelayStdDev;
 
     EXPECT_CALL(audio_processing_, GetStatistics())
         .WillRepeatedly(Return(audio_processing_stats_));
   }
 
+  void SyncWorkerQueue() {
+    static const size_t kDefaultTimeoutMs = 150;
+    rtc::Event event(false, false);
+    worker_queue_.PostTask([this, &event] { event.Set(); });
+    EXPECT_TRUE(event.Wait(kDefaultTimeoutMs));
+  }
+
  private:
   SimulatedClock simulated_clock_;
   testing::StrictMock<MockVoiceEngine> voice_engine_;
   rtc::scoped_refptr<AudioState> audio_state_;
   AudioSendStream::Config stream_config_;
   testing::StrictMock<MockVoEChannelProxy>* channel_proxy_ = nullptr;
   testing::NiceMock<MockCongestionObserver> bitrate_observer_;
   testing::NiceMock<MockRemoteBitrateObserver> remote_bitrate_observer_;
   MockAudioProcessing audio_processing_;
   AudioProcessing::AudioProcessingStatistics audio_processing_stats_;
@@ skipping 182 matching lines @@
 }
 
 TEST(AudioSendStreamTest, DoesNotPassHigherBitrateThanMaxBitrate) {
   ConfigHelper helper;
   internal::AudioSendStream send_stream(
       helper.config(), helper.audio_state(), helper.worker_queue(),
       helper.packet_router(), helper.congestion_controller(),
       helper.bitrate_allocator(), helper.event_log(), helper.rtcp_rtt_stats());
   EXPECT_CALL(*helper.channel_proxy(),
               SetBitrate(helper.config().max_bitrate_bps, _));
+  EXPECT_CALL(*helper.channel_proxy(), AdaptCodec());
   send_stream.OnBitrateUpdated(helper.config().max_bitrate_bps + 5000, 0.0, 50,
                                6000);
+  helper.SyncWorkerQueue();
 }
 
 TEST(AudioSendStreamTest, ProbingIntervalOnBitrateUpdated) {
   ConfigHelper helper;
   internal::AudioSendStream send_stream(
       helper.config(), helper.audio_state(), helper.worker_queue(),
       helper.packet_router(), helper.congestion_controller(),
       helper.bitrate_allocator(), helper.event_log(), helper.rtcp_rtt_stats());
   EXPECT_CALL(*helper.channel_proxy(), SetBitrate(_, 5000));
+  EXPECT_CALL(*helper.channel_proxy(), AdaptCodec());
   send_stream.OnBitrateUpdated(50000, 0.0, 50, 5000);
+  helper.SyncWorkerQueue();
 }
 
 }  // namespace test
 }  // namespace webrtc