Dead code elimination: scythe.chrome_functions:segment.path %media% edition, round 1.

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 "base/environment.h"
 #include "base/test/test_timeouts.h"
 #include "content/renderer/media/audio_hardware.h"
 #include "content/renderer/media/webrtc_audio_device_impl.h"
 #include "content/test/webrtc_audio_device_test.h"
 #include "media/audio/audio_manager.h"
 #include "media/audio/audio_util.h"
 #include "testing/gmock/include/gmock/gmock.h"
 #include "third_party/webrtc/voice_engine/main/interface/voe_audio_processing.h"
 #include "third_party/webrtc/voice_engine/main/interface/voe_base.h"
 #include "third_party/webrtc/voice_engine/main/interface/voe_external_media.h"
 #include "third_party/webrtc/voice_engine/main/interface/voe_file.h"
 #include "third_party/webrtc/voice_engine/main/interface/voe_network.h"
 
 using testing::_;
 using testing::AnyNumber;
 using testing::InvokeWithoutArgs;
 using testing::Return;
 using testing::StrEq;
 
 namespace {
 
-ACTION_P(QuitMessageLoop, loop_or_proxy) {
-  loop_or_proxy->PostTask(FROM_HERE, MessageLoop::QuitClosure());
-}
-
 class AudioUtil : public AudioUtilInterface {
  public:
   AudioUtil() {}
 
   virtual int GetAudioHardwareSampleRate() OVERRIDE {
     return media::GetAudioHardwareSampleRate();
   }
   virtual int GetAudioInputHardwareSampleRate(
       const std::string& device_id) OVERRIDE {
     return media::GetAudioInputHardwareSampleRate(device_id);
@@ skipping 119 matching lines @@
   int packet_size() const {
     base::AutoLock auto_lock(lock_);
     return packet_size_;
   }
 
   int sample_rate() const {
     base::AutoLock auto_lock(lock_);
     return sample_rate_;
   }
 
-  int channels() const {
-    base::AutoLock auto_lock(lock_);
-    return channels_;
-  }
-
  private:
   base::WaitableEvent* event_;
   int channel_id_;
   webrtc::ProcessingTypes type_;
   int packet_size_;
   int sample_rate_;
   int channels_;
   mutable base::Lock lock_;
   DISALLOW_COPY_AND_ASSIGN(WebRTCMediaProcessImpl);
 };
@@ skipping 335 matching lines @@
                                 MessageLoop::QuitClosure(),
                                 TestTimeouts::action_timeout());
   message_loop_.Run();
 
   EXPECT_EQ(0, base->StopSend(ch));
   EXPECT_EQ(0, base->StopPlayout(ch));
 
   EXPECT_EQ(0, base->DeleteChannel(ch));
   EXPECT_EQ(0, base->Terminate());
 }