Revert of Forward output glitch information from stream WebRTC log

media/audio/win/audio_low_latency_output_win_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 "media/audio/win/audio_low_latency_output_win.h"
 
 #include <windows.h>
 #include <mmsystem.h>
 #include <stddef.h>
 #include <stdint.h>
@@ skipping 191 matching lines @@
   int channels() const { return ChannelLayoutToChannelCount(channel_layout_); }
   int bits_per_sample() const { return bits_per_sample_; }
   int sample_rate() const { return sample_rate_; }
   int samples_per_packet() const { return samples_per_packet_; }
 
  private:
   AudioOutputStream* CreateOutputStream() {
     AudioOutputStream* aos = audio_man_->MakeAudioOutputStream(
         AudioParameters(format_, channel_layout_, sample_rate_,
                         bits_per_sample_, samples_per_packet_),
-        std::string(), AudioManager::LogCallback());
+        std::string());
     EXPECT_TRUE(aos);
     return aos;
   }
 
   AudioManager* audio_man_;
   AudioParameters::Format format_;
   ChannelLayout channel_layout_;
   int bits_per_sample_;
   int sample_rate_;
   int samples_per_packet_;
@@ skipping 402 matching lines @@
   aos->Start(&source);
   message_loop_.PostDelayedTask(FROM_HERE,
                                 base::MessageLoop::QuitWhenIdleClosure(),
                                 TestTimeouts::action_timeout());
   message_loop_.Run();
   aos->Stop();
   aos->Close();
 }
 
 }  // namespace media