Forward output glitch information from stream WebRTC log

media/audio/audio_low_latency_input_output_unittest.cc

 // Copyright (c) 2012 The Chromium Authors. All rights reserved.
 // Use of this source code is governed by a BSD-style license that can be
 // found in the LICENSE file.
 
 #include <stddef.h>
 #include <stdint.h>
 
 #include <memory>
 
 #include "base/environment.h"
@@ skipping 283 matching lines @@
  public:
   typedef AudioOutputStream StreamType;
 
   static AudioParameters GetDefaultAudioStreamParameters(
       AudioManager* audio_manager) {
     return audio_manager->GetDefaultOutputStreamParameters();
   }
 
   static StreamType* CreateStream(AudioManager* audio_manager,
       const AudioParameters& params) {
-    return audio_manager->MakeAudioOutputStream(params, std::string());
+    return audio_manager->MakeAudioOutputStream(
+        params, std::string(), AudioManager::StatisticsCallback());
   }
 };
 
 // Traits template holding a trait of StreamType. It encapsulates
 // AudioInputStream and AudioOutputStream stream types.
 template <typename StreamTraits>
 class StreamWrapper {
  public:
   typedef typename StreamTraits::StreamType StreamType;
 
@@ skipping 119 matching lines @@
 
   // All Close() operations that run on the mocked audio thread,
   // should be synchronous and not post additional close tasks to
   // mocked the audio thread. Hence, there is no need to call
   // message_loop()->RunUntilIdle() after the Close() methods.
   aos->Close();
   ais->Close();
 }
 
 }  // namespace media