Avoid reporting 0 as input level when AudioProcessing zero out the audio data

content/renderer/media/webrtc/webrtc_local_audio_track_adapter_unittest.cc

Index: content/renderer/media/webrtc/webrtc_local_audio_track_adapter_unittest.cc
diff --git a/content/renderer/media/webrtc/webrtc_local_audio_track_adapter_unittest.cc b/content/renderer/media/webrtc/webrtc_local_audio_track_adapter_unittest.cc
index f4a037ca6e97e7c3f22117c24bc3e0b7a9c55ea1..c7afce616effe21d7ce227b83b4493f73144d21f 100644
--- a/content/renderer/media/webrtc/webrtc_local_audio_track_adapter_unittest.cc
+++ b/content/renderer/media/webrtc/webrtc_local_audio_track_adapter_unittest.cc
@@ -74,14 +74,14 @@ TEST_F(WebRtcLocalAudioTrackAdapterTest, AddAndRemoveSink) {
   EXPECT_CALL(*sink,
               OnData(_, 16, params_.sample_rate(), params_.channels(),
                      params_.frames_per_buffer()));
-  track_->Capture(data.get(), base::TimeDelta(), 255, false, false);
+  track_->Capture(data.get(), base::TimeDelta(), 255, false, false, false);
 
   // Remove the sink from the webrtc track.
   webrtc_track->RemoveSink(sink.get());
   sink.reset();
 
   // Verify that no more callback gets into the sink.
-  track_->Capture(data.get(), base::TimeDelta(), 255, false, false);
+  track_->Capture(data.get(), base::TimeDelta(), 255, false, false, false);
 }
 
 TEST_F(WebRtcLocalAudioTrackAdapterTest, GetSignalLevel) {