Data race fix: Replace use of MessageLoop raw pointer in MSAudioProcessor.

content/renderer/media/media_stream_audio_processor.h

 // Copyright 2013 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_MEDIA_STREAM_AUDIO_PROCESSOR_H_
 #define CONTENT_RENDERER_MEDIA_MEDIA_STREAM_AUDIO_PROCESSOR_H_
 
 #include "base/atomicops.h"
 #include "base/files/file.h"
 #include "base/gtest_prod_util.h"
 #include "base/macros.h"
+#include "base/memory/ref_counted.h"
+#include "base/single_thread_task_runner.h"
 #include "base/synchronization/lock.h"
 #include "base/threading/thread_checker.h"
 #include "base/time/time.h"
 #include "content/common/content_export.h"
 #include "content/public/common/media_stream_request.h"
 #include "content/renderer/media/aec_dump_message_filter.h"
 #include "content/renderer/media/audio_repetition_detector.h"
 #include "content/renderer/media/webrtc_audio_device_impl.h"
 #include "media/base/audio_converter.h"
 #include "third_party/webrtc/api/mediastreaminterface.h"
@@ skipping 27 matching lines @@
 // on the getUserMedia constraints, processes the data and outputs it in a unit
 // of 10 ms data chunk.
 class CONTENT_EXPORT MediaStreamAudioProcessor :
     NON_EXPORTED_BASE(public WebRtcPlayoutDataSource::Sink),
     NON_EXPORTED_BASE(public AudioProcessorInterface),
     NON_EXPORTED_BASE(public AecDumpMessageFilter::AecDumpDelegate) {
  public:
   // |playout_data_source| is used to register this class as a sink to the
   // WebRtc playout data for processing AEC. If clients do not enable AEC,
   // |playout_data_source| won't be used.
+  //
+  // Threading note: The constructor assumes it is being run on the main render
+  // thread.
   MediaStreamAudioProcessor(
       const blink::WebMediaConstraints& constraints,
       const MediaStreamDevice::AudioDeviceParameters& input_params,
       WebRtcPlayoutDataSource* playout_data_source);
 
   // Called when the format of the capture data has changed.
   // Called on the main render thread. The caller is responsible for stopping
   // the capture thread before calling this method.
   // After this method, the capture thread will be changed to a new capture
   // thread.
@@ skipping 121 matching lines @@
   // capture audio thread.
   media::AudioParameters input_format_;
   media::AudioParameters output_format_;
   // Only used on the render audio thread.
   media::AudioParameters render_format_;
 
   // Raw pointer to the WebRtcPlayoutDataSource, which is valid for the
   // lifetime of RenderThread.
   WebRtcPlayoutDataSource* playout_data_source_;
 
-  // Used to DCHECK that some methods are called on the main render thread.
-  base::ThreadChecker main_thread_checker_;
+  // Task runner for the main render thread.
+  const scoped_refptr<base::SingleThreadTaskRunner> main_thread_runner_;
+
   // Used to DCHECK that some methods are called on the capture audio thread.
   base::ThreadChecker capture_thread_checker_;
   // Used to DCHECK that some methods are called on the render audio thread.
   base::ThreadChecker render_thread_checker_;
 
-  // Message loop for the main render thread. We're assuming that we're created
-  // on the main render thread.
-  base::MessageLoop* main_thread_message_loop_;
-
   // Flag to enable stereo channel mirroring.
   bool audio_mirroring_;
 
   // Typing detector. |typing_detected_| is used to show the result of typing
   // detection. It can be accessed by the capture audio thread and by the
   // libjingle thread which calls GetStats().
   std::unique_ptr<webrtc::TypingDetection> typing_detector_;
   base::subtle::Atomic32 typing_detected_;
 
   // Communication with browser for AEC dump.
   scoped_refptr<AecDumpMessageFilter> aec_dump_message_filter_;
 
   // Flag to avoid executing Stop() more than once.
   bool stopped_;
 
   // Object for logging UMA stats for echo information when the AEC is enabled.
   // Accessed on the main render thread.
   std::unique_ptr<EchoInformation> echo_information_;
 
   DISALLOW_COPY_AND_ASSIGN(MediaStreamAudioProcessor);
 };
 
 }  // namespace content
 
 #endif  // CONTENT_RENDERER_MEDIA_MEDIA_STREAM_AUDIO_PROCESSOR_H_