Forward output glitch information from stream WebRTC log

chromecast/media/audio/cast_audio_output_stream_unittest.cc

 // Copyright 2015 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 "chromecast/media/audio/cast_audio_output_stream.h"
 
 #include "base/run_loop.h"
 #include "chromecast/base/metrics/cast_metrics_test_helper.h"
 #include "chromecast/media/audio/cast_audio_manager.h"
 #include "chromecast/public/media/cast_decoder_buffer.h"
@@ skipping 217 matching lines @@
   FakeMediaPipelineBackend* GetBackend() {
     return audio_manager_->media_pipeline_backend();
   }
 
   FakeAudioDecoder* GetAudio() {
     FakeMediaPipelineBackend* backend = GetBackend();
     return (backend ? backend->decoder() : nullptr);
   }
 
   ::media::AudioOutputStream* CreateStream() {
-    return audio_manager_->MakeAudioOutputStream(GetAudioParams(),
-                                                 kDefaultDeviceId);
+    return audio_manager_->MakeAudioOutputStream(
+        GetAudioParams(), kDefaultDeviceId,
+        ::media::AudioManager::LogCallback());
   }
 
   // Runs the messsage loop for duration equivalent to the given number of
   // audio |frames|.
   void RunMessageLoopFor(int frames) {
     ::media::AudioParameters audio_params = GetAudioParams();
     base::TimeDelta duration = audio_params.GetBufferDuration() * frames;
 
     base::RunLoop run_loop;
     message_loop_.task_runner()->PostDelayedTask(
@@ skipping 263 matching lines @@
 TEST_F(CastAudioOutputStreamTest, CloseWithoutStart) {
   ::media::AudioOutputStream* stream = CreateStream();
   ASSERT_TRUE(stream);
   ASSERT_TRUE(stream->Open());
   stream->Close();
 }
 
 }  // namespace
 }  // namespace media
 }  // namespace chromecast