Replace LOG(INFO) with VLOG(0), throughout *media*

content/renderer/media/webrtc_audio_device_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 <vector>
 
 #include "base/environment.h"
 #include "base/file_util.h"
 #include "base/files/file_path.h"
 #include "base/path_service.h"
@@ skipping 827 matching lines @@
   EXPECT_EQ(0, base->StartPlayout(ch));
   EXPECT_EQ(0, base->StartSend(ch));
   scoped_refptr<WebRtcAudioRenderer> renderer(
       CreateDefaultWebRtcAudioRenderer(kRenderViewId));
   scoped_refptr<MediaStreamAudioRenderer> proxy(
       renderer->CreateSharedAudioRendererProxy());
   EXPECT_TRUE(webrtc_audio_device->SetAudioRenderer(renderer.get()));
   proxy->Start();
   proxy->Play();
 
-  LOG(INFO) << ">> You should now be able to hear yourself in loopback...";
+  VLOG(0) << ">> You should now be able to hear yourself in loopback...";
   message_loop_.PostDelayedTask(FROM_HERE,
                                 base::MessageLoop::QuitClosure(),
                                 base::TimeDelta::FromSeconds(2));
   message_loop_.Run();
 
   webrtc_audio_device->GetDefaultCapturer()->Stop();
   proxy->Stop();
   EXPECT_EQ(0, base->StopSend(ch));
   EXPECT_EQ(0, base->StopPlayout(ch));
 
@@ skipping 149 matching lines @@
     LOG(WARNING) << "Test disabled due to the test hangs on WinXP.";
     return;
   }
 #endif
   int latency = RunWebRtcLoopbackTimeTest(audio_manager_.get(), true);
   PrintPerfResultMs("webrtc_loopback_with_signal_processing (100 packets)",
                     "t", latency);
 }
 
 }  // namespace content