Log audio system used to WebRTC log.

media/audio/audio_output_proxy_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 <stdint.h>
 
 #include <memory>
 #include <string>
 
 #include "base/message_loop/message_loop.h"
@@ skipping 98 matching lines @@
   MOCK_METHOD3(MakeAudioInputStream,
                AudioInputStream*(const AudioParameters& params,
                                  const std::string& device_id,
                                  const LogCallback& log_callback));
   MOCK_METHOD0(ShowAudioInputSettings, void());
   MOCK_METHOD0(GetTaskRunner, scoped_refptr<base::SingleThreadTaskRunner>());
   MOCK_METHOD0(GetWorkerTaskRunner,
                scoped_refptr<base::SingleThreadTaskRunner>());
   MOCK_METHOD1(GetAudioInputDeviceNames, void(
       media::AudioDeviceNames* device_name));
+  MOCK_METHOD0(GetName, const char*());
 
   MOCK_METHOD2(MakeLinearOutputStream,
                AudioOutputStream*(const AudioParameters& params,
                                   const LogCallback& log_callback));
   MOCK_METHOD3(MakeLowLatencyOutputStream,
                AudioOutputStream*(const AudioParameters& params,
                                   const std::string& device_id,
                                   const LogCallback& log_callback));
   MOCK_METHOD3(MakeLinearInputStream,
                AudioInputStream*(const AudioParameters& params,
@@ skipping 612 matching lines @@
       .WillOnce(Return(&real_stream));
 
   // Stream1 should be able to successfully open and start.
   EXPECT_CALL(real_stream, Open()).WillOnce(Return(true));
   proxy = new AudioOutputProxy(resampler_.get());
   EXPECT_TRUE(proxy->Open());
   CloseAndWaitForCloseTimer(proxy, &real_stream);
 }
 
 }  // namespace media