Deprecate NonThreadSafe in content/renderer/media in favor of SequenceChecker.

content/renderer/media/webrtc_audio_renderer.h

 // 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.
 
 #ifndef CONTENT_RENDERER_MEDIA_WEBRTC_AUDIO_RENDERER_H_
 #define CONTENT_RENDERER_MEDIA_WEBRTC_AUDIO_RENDERER_H_
 
 #include <stdint.h>
 
 #include <map>
 #include <memory>
 #include <string>
 #include <vector>
 
 #include "base/macros.h"
 #include "base/memory/ref_counted.h"
+#include "base/sequence_checker.h"
 #include "base/single_thread_task_runner.h"
 #include "base/synchronization/lock.h"
-#include "base/threading/non_thread_safe.h"
 #include "base/threading/thread_checker.h"
 #include "content/public/renderer/media_stream_audio_renderer.h"
 #include "content/renderer/media/webrtc_audio_device_impl.h"
 #include "media/base/audio_decoder.h"
 #include "media/base/audio_pull_fifo.h"
 #include "media/base/audio_renderer_sink.h"
 #include "media/base/channel_layout.h"
 #include "third_party/WebKit/public/platform/WebMediaStream.h"
 
 namespace webrtc {
 class AudioSourceInterface;
 }  // namespace webrtc
 
 namespace content {
 
 class WebRtcAudioRendererSource;
 
 // This renderer handles calls from the pipeline and WebRtc ADM. It is used
 // for connecting WebRtc MediaStream with the audio pipeline.
 class CONTENT_EXPORT WebRtcAudioRenderer
     : NON_EXPORTED_BASE(public media::AudioRendererSink::RenderCallback),
       NON_EXPORTED_BASE(public MediaStreamAudioRenderer) {
  public:
   // This is a little utility class that holds the configured state of an audio
   // stream.
   // It is used by both WebRtcAudioRenderer and SharedAudioRenderer (see cc
   // file) so a part of why it exists is to avoid code duplication and track
   // the state in the same way in WebRtcAudioRenderer and SharedAudioRenderer.
-  class PlayingState : public base::NonThreadSafe {
+  class PlayingState {
    public:
     PlayingState() : playing_(false), volume_(1.0f) {}
 
+    ~PlayingState() { DCHECK_CALLED_ON_VALID_SEQUENCE(sequence_checker_); }
+
     bool playing() const {
-      DCHECK(CalledOnValidThread());
+      DCHECK_CALLED_ON_VALID_SEQUENCE(sequence_checker_);
       return playing_;
     }
 
     void set_playing(bool playing) {
-      DCHECK(CalledOnValidThread());
+      DCHECK_CALLED_ON_VALID_SEQUENCE(sequence_checker_);
       playing_ = playing;
     }
 
     float volume() const {
-      DCHECK(CalledOnValidThread());
+      DCHECK_CALLED_ON_VALID_SEQUENCE(sequence_checker_);
       return volume_;
     }
 
     void set_volume(float volume) {
-      DCHECK(CalledOnValidThread());
+      DCHECK_CALLED_ON_VALID_SEQUENCE(sequence_checker_);
       volume_ = volume;
     }
 
    private:
     bool playing_;
     float volume_;
+
+    SEQUENCE_CHECKER(sequence_checker_);
   };
 
   WebRtcAudioRenderer(
       const scoped_refptr<base::SingleThreadTaskRunner>& signaling_thread,
       const blink::WebMediaStream& media_stream,
       int source_render_frame_id,
       int session_id,
       const std::string& device_id,
       const url::Origin& security_origin);
 
@@ skipping 173 matching lines @@
   // Stores the maximum time spent waiting for render data from the source. Used
   // for logging UMA data. Logged and reset when Stop() is called.
   base::TimeDelta max_render_time_;
 
   DISALLOW_IMPLICIT_CONSTRUCTORS(WebRtcAudioRenderer);
 };
 
 }  // namespace content
 
 #endif  // CONTENT_RENDERER_MEDIA_WEBRTC_AUDIO_RENDERER_H_