Rename TaskRunner::RunsTasksOnCurrentThread() in //content

content/renderer/media/webrtc/webrtc_audio_sink.cc

 // Copyright 2016 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 "content/renderer/media/webrtc/webrtc_audio_sink.h"
 
 #include <algorithm>
 #include <limits>
 
 #include "base/bind.h"
@@ skipping 121 matching lines @@
                  number_of_channels, number_of_frames);
   }
 }
 
 std::string WebRtcAudioSink::Adapter::kind() const {
   return webrtc::MediaStreamTrackInterface::kAudioKind;
 }
 
 bool WebRtcAudioSink::Adapter::set_enabled(bool enable) {
   DCHECK(!signaling_task_runner_ ||
-         signaling_task_runner_->RunsTasksOnCurrentThread());
+         signaling_task_runner_->RunsTasksInCurrentSequence());
   return webrtc::MediaStreamTrack<webrtc::AudioTrackInterface>::
       set_enabled(enable);
 }
 
 void WebRtcAudioSink::Adapter::AddSink(webrtc::AudioTrackSinkInterface* sink) {
   DCHECK(!signaling_task_runner_ ||
-         signaling_task_runner_->RunsTasksOnCurrentThread());
+         signaling_task_runner_->RunsTasksInCurrentSequence());
   DCHECK(sink);
   base::AutoLock auto_lock(lock_);
   DCHECK(std::find(sinks_.begin(), sinks_.end(), sink) == sinks_.end());
   sinks_.push_back(sink);
 }
 
 void WebRtcAudioSink::Adapter::RemoveSink(
     webrtc::AudioTrackSinkInterface* sink) {
   DCHECK(!signaling_task_runner_ ||
-         signaling_task_runner_->RunsTasksOnCurrentThread());
+         signaling_task_runner_->RunsTasksInCurrentSequence());
   base::AutoLock auto_lock(lock_);
   const auto it = std::find(sinks_.begin(), sinks_.end(), sink);
   if (it != sinks_.end())
     sinks_.erase(it);
 }
 
 bool WebRtcAudioSink::Adapter::GetSignalLevel(int* level) {
   DCHECK(!signaling_task_runner_ ||
-         signaling_task_runner_->RunsTasksOnCurrentThread());
+         signaling_task_runner_->RunsTasksInCurrentSequence());
 
   // |level_| is only set once, so it's safe to read without first acquiring a
   // mutex.
   if (!level_)
     return false;
   const float signal_level = level_->GetCurrent();
   DCHECK_GE(signal_level, 0.0f);
   DCHECK_LE(signal_level, 1.0f);
   // Convert from float in range [0.0,1.0] to an int in range [0,32767].
   *level = static_cast<int>(signal_level * std::numeric_limits<int16_t>::max() +
                             0.5f /* rounding to nearest int */);
   return true;
 }
 
 rtc::scoped_refptr<webrtc::AudioProcessorInterface>
 WebRtcAudioSink::Adapter::GetAudioProcessor() {
   DCHECK(!signaling_task_runner_ ||
-         signaling_task_runner_->RunsTasksOnCurrentThread());
+         signaling_task_runner_->RunsTasksInCurrentSequence());
   return audio_processor_.get();
 }
 
 webrtc::AudioSourceInterface* WebRtcAudioSink::Adapter::GetSource() const {
   DCHECK(!signaling_task_runner_ ||
-         signaling_task_runner_->RunsTasksOnCurrentThread());
+         signaling_task_runner_->RunsTasksInCurrentSequence());
   return source_.get();
 }
 
 }  // namespace content