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

content/renderer/media/media_stream_audio_processor.cc

 // 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.
 
 #include "content/renderer/media/media_stream_audio_processor.h"
 
 #include <stddef.h>
 #include <stdint.h>
 #include <utility>
 
 #include "base/command_line.h"
 #include "base/metrics/field_trial.h"
 #include "base/metrics/histogram_macros.h"
 #include "base/single_thread_task_runner.h"
 #include "base/strings/string_number_conversions.h"
+#include "base/threading/thread_task_runner_handle.h"
 #include "base/trace_event/trace_event.h"
 #include "build/build_config.h"
 #include "content/public/common/content_switches.h"
 #include "content/renderer/media/media_stream_audio_processor_options.h"
 #include "content/renderer/media/webrtc_audio_device_impl.h"
 #include "media/base/audio_converter.h"
 #include "media/base/audio_fifo.h"
 #include "media/base/audio_parameters.h"
 #include "media/base/channel_layout.h"
 #include "third_party/WebKit/public/platform/WebMediaConstraints.h"
@@ skipping 255 matching lines @@
   // to Consume().  Only used when the FIFO is disabled.
   bool data_available_;
 };
 
 MediaStreamAudioProcessor::MediaStreamAudioProcessor(
     const blink::WebMediaConstraints& constraints,
     const MediaStreamDevice::AudioDeviceParameters& input_params,
     WebRtcPlayoutDataSource* playout_data_source)
     : render_delay_ms_(0),
       playout_data_source_(playout_data_source),
-      main_thread_message_loop_(base::MessageLoop::current()),
+      main_thread_runner_(base::ThreadTaskRunnerHandle::Get()),
       audio_mirroring_(false),
       typing_detected_(false),
       stopped_(false) {
+  DCHECK(main_thread_runner_);
   capture_thread_checker_.DetachFromThread();
   render_thread_checker_.DetachFromThread();
   InitializeAudioProcessingModule(constraints, input_params);
 
   aec_dump_message_filter_ = AecDumpMessageFilter::Get();
   // In unit tests not creating a message filter, |aec_dump_message_filter_|
   // will be NULL. We can just ignore that. Other unit tests and browser tests
   // ensure that we do get the filter when we should.
   if (aec_dump_message_filter_.get())
     aec_dump_message_filter_->AddDelegate(this);
 
   // Create and configure |audio_repetition_detector_|.
   std::vector<int> look_back_times;
   for (int time = kMaxLookbackTimeMs; time >= kMinLookbackTimeMs;
        time -= kLookbackTimeStepMs) {
     look_back_times.push_back(time);
   }
   audio_repetition_detector_.reset(
       new AudioRepetitionDetector(kMinLengthMs, kMaxFrames, look_back_times,
                                   base::Bind(&ReportRepetition)));
 }
 
 MediaStreamAudioProcessor::~MediaStreamAudioProcessor() {
   // TODO(miu): This class is ref-counted, shared among threads, and then
   // requires itself to be destroyed on the main thread only?!?!? Fix this, and
   // then remove the hack in WebRtcAudioSink::Adapter.
-  DCHECK(main_thread_checker_.CalledOnValidThread());
+  DCHECK(main_thread_runner_->BelongsToCurrentThread());
   Stop();
 }
 
 void MediaStreamAudioProcessor::OnCaptureFormatChanged(
     const media::AudioParameters& input_format) {
-  DCHECK(main_thread_checker_.CalledOnValidThread());
+  DCHECK(main_thread_runner_->BelongsToCurrentThread());
+
   // There is no need to hold a lock here since the caller guarantees that
   // there is no more PushCaptureData() and ProcessAndConsumeData() callbacks
   // on the capture thread.
   InitializeCaptureFifo(input_format);
 
   // Reset the |capture_thread_checker_| since the capture data will come from
   // a new capture thread.
   capture_thread_checker_.DetachFromThread();
 }
 
@@ skipping 45 matching lines @@
     // Swap the first and second channels.
     output_bus->bus()->SwapChannels(0, 1);
   }
 
   *processed_data = output_bus->bus();
 
   return true;
 }
 
 void MediaStreamAudioProcessor::Stop() {
-  DCHECK(main_thread_checker_.CalledOnValidThread());
+  DCHECK(main_thread_runner_->BelongsToCurrentThread());
 
   if (stopped_)
     return;
 
   stopped_ = true;
 
   if (aec_dump_message_filter_.get()) {
     aec_dump_message_filter_->RemoveDelegate(this);
     aec_dump_message_filter_ = NULL;
   }
@@ skipping 16 matching lines @@
 const media::AudioParameters& MediaStreamAudioProcessor::InputFormat() const {
   return input_format_;
 }
 
 const media::AudioParameters& MediaStreamAudioProcessor::OutputFormat() const {
   return output_format_;
 }
 
 void MediaStreamAudioProcessor::OnAecDumpFile(
     const IPC::PlatformFileForTransit& file_handle) {
-  DCHECK(main_thread_checker_.CalledOnValidThread());
+  DCHECK(main_thread_runner_->BelongsToCurrentThread());
 
   base::File file = IPC::PlatformFileForTransitToFile(file_handle);
   DCHECK(file.IsValid());
 
   if (audio_processing_)
     StartEchoCancellationDump(audio_processing_.get(), std::move(file));
   else
     file.Close();
 }
 
 void MediaStreamAudioProcessor::OnDisableAecDump() {
-  DCHECK(main_thread_checker_.CalledOnValidThread());
+  DCHECK(main_thread_runner_->BelongsToCurrentThread());
   if (audio_processing_)
     StopEchoCancellationDump(audio_processing_.get());
 }
 
 void MediaStreamAudioProcessor::OnIpcClosing() {
-  DCHECK(main_thread_checker_.CalledOnValidThread());
+  DCHECK(main_thread_runner_->BelongsToCurrentThread());
   aec_dump_message_filter_ = NULL;
 }
 
 // static
 bool MediaStreamAudioProcessor::WouldModifyAudio(
     const blink::WebMediaConstraints& constraints,
     int effects_flags) {
   // Note: This method should by kept in-sync with any changes to the logic in
   // MediaStreamAudioProcessor::InitializeAudioProcessingModule().
 
@@ skipping 54 matching lines @@
     // TODO(ajm): Should AnalyzeReverseStream() account for the |audio_delay|?
     audio_processing_->AnalyzeReverseStream(
         analysis_bus->channel_ptrs(),
         analysis_bus->bus()->frames(),
         sample_rate,
         ChannelsToLayout(audio_bus->channels()));
   }
 }
 
 void MediaStreamAudioProcessor::OnPlayoutDataSourceChanged() {
-  DCHECK(main_thread_checker_.CalledOnValidThread());
+  DCHECK(main_thread_runner_->BelongsToCurrentThread());
   // There is no need to hold a lock here since the caller guarantees that
   // there is no more OnPlayoutData() callback on the render thread.
   render_thread_checker_.DetachFromThread();
   render_fifo_.reset();
 }
 
 void MediaStreamAudioProcessor::OnRenderThreadChanged() {
   render_thread_checker_.DetachFromThread();
   DCHECK(render_thread_checker_.CalledOnValidThread());
   render_fifo_->ReattachThreadChecker();
 }
 
 void MediaStreamAudioProcessor::GetStats(AudioProcessorStats* stats) {
   stats->typing_noise_detected =
       (base::subtle::Acquire_Load(&typing_detected_) != false);
   GetAecStats(audio_processing_.get()->echo_cancellation(), stats);
 }
 
 void MediaStreamAudioProcessor::InitializeAudioProcessingModule(
     const blink::WebMediaConstraints& constraints,
     const MediaStreamDevice::AudioDeviceParameters& input_params) {
-  DCHECK(main_thread_checker_.CalledOnValidThread());
+  DCHECK(main_thread_runner_->BelongsToCurrentThread());
   DCHECK(!audio_processing_);
 
   MediaAudioConstraints audio_constraints(constraints, input_params.effects);
 
   // Note: The audio mirroring constraint (i.e., swap left and right channels)
   // is handled within this MediaStreamAudioProcessor and does not, by itself,
   // require webrtc::AudioProcessing.
   audio_mirroring_ = audio_constraints.GetGoogAudioMirroring();
 
   const bool echo_cancellation =
@@ skipping 98 matching lines @@
   }
 
   if (goog_agc)
     EnableAutomaticGainControl(audio_processing_.get());
 
   RecordProcessingState(AUDIO_PROCESSING_ENABLED);
 }
 
 void MediaStreamAudioProcessor::InitializeCaptureFifo(
     const media::AudioParameters& input_format) {
-  DCHECK(main_thread_checker_.CalledOnValidThread());
+  DCHECK(main_thread_runner_->BelongsToCurrentThread());
   DCHECK(input_format.IsValid());
   input_format_ = input_format;
 
   // TODO(ajm): For now, we assume fixed parameters for the output when audio
   // processing is enabled, to match the previous behavior. We should either
   // use the input parameters (in which case, audio processing will convert
   // at output) or ideally, have a backchannel from the sink to know what
   // format it would prefer.
 #if defined(OS_ANDROID)
   int audio_processing_sample_rate = AudioProcessing::kSampleRate16kHz;
@@ skipping 127 matching lines @@
   DCHECK_EQ(err, 0) << "ProcessStream() error: " << err;
 
   if (typing_detector_) {
     webrtc::VoiceDetection* vad = ap->voice_detection();
     DCHECK(vad->is_enabled());
     bool detected = typing_detector_->Process(key_pressed,
                                               vad->stream_has_voice());
     base::subtle::Release_Store(&typing_detected_, detected);
   }
 
-  main_thread_message_loop_->task_runner()->PostTask(
+  main_thread_runner_->PostTask(

[miu, 2016/10/19 02:06:40]

Note: This LOC is where the data race occurred.

       FROM_HERE, base::Bind(&MediaStreamAudioProcessor::UpdateAecStats, this));
 
   // Return 0 if the volume hasn't been changed, and otherwise the new volume.
   return (agc->stream_analog_level() == volume) ?
       0 : agc->stream_analog_level();
 }
 
 void MediaStreamAudioProcessor::UpdateAecStats() {
-  DCHECK(main_thread_checker_.CalledOnValidThread());
+  DCHECK(main_thread_runner_->BelongsToCurrentThread());
   if (echo_information_)
     echo_information_->UpdateAecStats(audio_processing_->echo_cancellation());
 }
 
 }  // namespace content