Update smoothed bitrate.

webrtc/audio/audio_send_stream_unittest.cc

Index: webrtc/audio/audio_send_stream_unittest.cc
diff --git a/webrtc/audio/audio_send_stream_unittest.cc b/webrtc/audio/audio_send_stream_unittest.cc
index 4efd91b3e5618e2c6e5e4951836c9c124aebce56..e760b33dd03d6652f6271eeb7e897bd7ea2fc1b7 100644
--- a/webrtc/audio/audio_send_stream_unittest.cc
+++ b/webrtc/audio/audio_send_stream_unittest.cc
@@ -217,6 +217,13 @@ struct ConfigHelper {
         .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_;
@@ -419,8 +426,10 @@ TEST(AudioSendStreamTest, DoesNotPassHigherBitrateThanMaxBitrate) {
       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) {
@@ -430,7 +439,9 @@ TEST(AudioSendStreamTest, ProbingIntervalOnBitrateUpdated) {
       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