Forward output glitch information from stream WebRTC log

media/audio/win/audio_low_latency_input_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_input_win.h"
 
 #include <windows.h>
 #include <mmsystem.h>
 #include <stddef.h>
 #include <stdint.h>
@@ skipping 184 matching lines @@
   }
   int bits_per_sample() const { return default_params_.bits_per_sample(); }
   int sample_rate() const { return default_params_.sample_rate(); }
   int frames_per_buffer() const { return frames_per_buffer_; }
 
  private:
   AudioInputStream* CreateInputStream() {
     AudioParameters params = default_params_;
     params.set_frames_per_buffer(frames_per_buffer_);
     AudioInputStream* ais = audio_man_->MakeAudioInputStream(
-        params, AudioDeviceDescription::kDefaultDeviceId);
+        params, AudioDeviceDescription::kDefaultDeviceId,
+        AudioManager::LogCallback());
     EXPECT_TRUE(ais);
     return ais;
   }
 
   AudioManager* audio_man_;
   AudioParameters default_params_;
   int frames_per_buffer_;
 };
 
 // Convenience method which creates a default AudioInputStream object.
@@ skipping 217 matching lines @@
   AudioParameters params = audio_manager_->GetInputStreamParameters(
       AudioDeviceDescription::kLoopbackInputDeviceId);
   EXPECT_EQ(params.effects(), 0);
 
   AudioParameters output_params =
       audio_manager_->GetOutputStreamParameters(std::string());
   EXPECT_EQ(params.sample_rate(), output_params.sample_rate());
   EXPECT_EQ(params.channel_layout(), output_params.channel_layout());
 
   ScopedAudioInputStream stream(audio_manager_->MakeAudioInputStream(
-      params, AudioDeviceDescription::kLoopbackInputDeviceId));
+      params, AudioDeviceDescription::kLoopbackInputDeviceId,
+      AudioManager::LogCallback()));
   ASSERT_TRUE(stream->Open());
   FakeAudioInputCallback sink;
   stream->Start(&sink);
   ASSERT_FALSE(sink.error());
 
   sink.WaitForData();
   stream.Close();
 
   EXPECT_GT(sink.num_received_audio_frames(), 0);
   EXPECT_FALSE(sink.error());
@@ skipping 21 matching lines @@
   WriteToFileAudioSink file_sink(file_name, aisw.bits_per_sample());
   VLOG(0) << ">> Speak into the default microphone while recording.";
   ais->Start(&file_sink);
   base::PlatformThread::Sleep(TestTimeouts::action_timeout());
   ais->Stop();
   VLOG(0) << ">> Recording has stopped.";
   ais.Close();
 }
 
 }  // namespace media