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

Internal-only site: http://go/videostack/engineering/dead-code-elimination

Review URL: https://chromiumcodereview.appspot.com/10837118

git-svn-id: svn://svn.chromium.org/chrome/trunk/src@150129 0039d316-1c4b-4281-b951-d872f2087c98

5 files changed, 1 insertions, 23 deletions

diff --git a/content/renderer/media/audio_message_filter_unittest.cc b/content/renderer/media/audio_message_filter_unittest.cc
index f39998a..63f244f 100644
--- a/content/renderer/media/audio_message_filter_unittest.cc
+++ b/content/renderer/media/audio_message_filter_unittest.cc
@@ -32,11 +32,6 @@ class MockAudioDelegate : public media::AudioOutputIPCDelegate {
 
   virtual void OnIPCClosed() OVERRIDE {}
 
-  virtual void OnVolume(double volume) {
-    volume_received_ = true;
-    volume_ = volume;
-  }
-
   void Reset() {
     state_changed_received_ = false;
     state_ = media::AudioOutputIPCDelegate::kError;
@@ -56,9 +51,6 @@ class MockAudioDelegate : public media::AudioOutputIPCDelegate {
   base::SharedMemoryHandle handle() { return handle_; }
   uint32 length() { return length_; }
 
-  bool volume_received() { return volume_received_; }
-  double volume() { return volume_; }
-
  private:
   bool state_changed_received_;
   media::AudioOutputIPCDelegate::State state_;
diff --git a/content/renderer/media/media_stream_impl.h b/content/renderer/media/media_stream_impl.h
index 4c3d62a..970a2f2 100644
--- a/content/renderer/media/media_stream_impl.h
+++ b/content/renderer/media/media_stream_impl.h
@@ -180,8 +180,6 @@ class CONTENT_EXPORT MediaStreamImpl
   bool EnsurePeerConnectionFactory();
   void CleanupPeerConnectionFactory();
 
-  PeerConnectionHandlerBase* FindPeerConnectionByStream(
-      const WebKit::WebMediaStreamDescriptor& stream);
   scoped_refptr<media::VideoDecoder> CreateLocalVideoDecoder(
       webrtc::MediaStreamInterface* stream,
       media::MessageLoopFactory* message_loop_factory);
diff --git a/content/renderer/media/video_capture_message_filter_unittest.cc b/content/renderer/media/video_capture_message_filter_unittest.cc
index f9dd1f6..46ecc4d 100644
--- a/content/renderer/media/video_capture_message_filter_unittest.cc
+++ b/content/renderer/media/video_capture_message_filter_unittest.cc
@@ -1,4 +1,4 @@
-// Copyright (c) 2011 The Chromium Authors. All rights reserved.
+// 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.
 
@@ -78,7 +78,6 @@ class MockVideoCaptureDelegate : public VideoCaptureMessageFilter::Delegate {
   bool device_info_receive() { return device_info_received_; }
   const media::VideoCaptureParams& received_device_info() { return params_; }
 
-  bool device_id_received() { return device_id_received_; }
   int32 device_id() { return device_id_; }
 
  private:
diff --git a/content/renderer/media/webrtc_audio_device_impl.h b/content/renderer/media/webrtc_audio_device_impl.h
index 823bf0c..89ae1c0 100644
--- a/content/renderer/media/webrtc_audio_device_impl.h
+++ b/content/renderer/media/webrtc_audio_device_impl.h
@@ -382,8 +382,6 @@ class CONTENT_EXPORT WebRtcAudioDeviceImpl
   int output_sample_rate() const {
     return output_audio_parameters_.sample_rate();
   }
-  int input_delay_ms() const { return input_delay_ms_; }
-  int output_delay_ms() const { return output_delay_ms_; }
   bool initialized() const { return initialized_; }
   bool playing() const { return playing_; }
   bool recording() const { return recording_; }
diff --git a/content/renderer/media/webrtc_audio_device_unittest.cc b/content/renderer/media/webrtc_audio_device_unittest.cc
index c3db6d7..2937d86 100644
--- a/content/renderer/media/webrtc_audio_device_unittest.cc
+++ b/content/renderer/media/webrtc_audio_device_unittest.cc
@@ -24,10 +24,6 @@ using testing::StrEq;
 
 namespace {
 
-ACTION_P(QuitMessageLoop, loop_or_proxy) {
-  loop_or_proxy->PostTask(FROM_HERE, MessageLoop::QuitClosure());
-}
-
 class AudioUtil : public AudioUtilInterface {
  public:
   AudioUtil() {}
@@ -167,11 +163,6 @@ class WebRTCMediaProcessImpl : public webrtc::VoEMediaProcess {
     return sample_rate_;
   }
 
-  int channels() const {
-    base::AutoLock auto_lock(lock_);
-    return channels_;
-  }
-
  private:
   base::WaitableEvent* event_;
   int channel_id_;