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

content/renderer/media/media_stream_audio_processor_unittest.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 <stddef.h>
 #include <stdint.h>
 
 #include <vector>
 
 #include "base/files/file_path.h"
 #include "base/files/file_util.h"
 #include "base/logging.h"
 #include "base/macros.h"
 #include "base/memory/aligned_memory.h"
+#include "base/message_loop/message_loop.h"
 #include "base/path_service.h"
+#include "base/threading/thread_task_runner_handle.h"
 #include "base/time/time.h"
 #include "build/build_config.h"
 #include "content/common/media/media_stream_options.h"
 #include "content/public/common/media_stream_request.h"
 #include "content/renderer/media/media_stream_audio_processor.h"
 #include "content/renderer/media/media_stream_audio_processor_options.h"
 #include "content/renderer/media/mock_constraint_factory.h"
 #include "media/base/audio_bus.h"
 #include "media/base/audio_parameters.h"
 #include "testing/gmock/include/gmock/gmock.h"
@@ skipping 164 matching lines @@
     EXPECT_FALSE(audio_processing->voice_detection()->is_enabled());
 #else
     EXPECT_TRUE(audio_processing->gain_control()->mode() ==
         webrtc::GainControl::kAdaptiveAnalog);
     EXPECT_TRUE(audio_processing->voice_detection()->is_enabled());
     EXPECT_TRUE(audio_processing->voice_detection()->likelihood() ==
         webrtc::VoiceDetection::kVeryLowLikelihood);
 #endif
   }
 
+  base::MessageLoop main_thread_message_loop_;
   media::AudioParameters params_;
   MediaStreamDevice::AudioDeviceParameters input_device_params_;
 };
 
 // Test crashing with ASAN on Android. crbug.com/468762
 #if defined(OS_ANDROID) && defined(ADDRESS_SANITIZER)
 #define MAYBE_WithAudioProcessing DISABLED_WithAudioProcessing
 #else
 #define MAYBE_WithAudioProcessing WithAudioProcessing
 #endif
 TEST_F(MediaStreamAudioProcessorTest, MAYBE_WithAudioProcessing) {
-  base::MessageLoop message_loop;
   MockConstraintFactory constraint_factory;
   scoped_refptr<WebRtcAudioDeviceImpl> webrtc_audio_device(
       new WebRtcAudioDeviceImpl());
   scoped_refptr<MediaStreamAudioProcessor> audio_processor(
       new rtc::RefCountedObject<MediaStreamAudioProcessor>(
           constraint_factory.CreateWebMediaConstraints(), input_device_params_,
           webrtc_audio_device.get()));
   EXPECT_TRUE(audio_processor->has_audio_processing());
   audio_processor->OnCaptureFormatChanged(params_);
   VerifyDefaultComponents(audio_processor.get());
@@ skipping 214 matching lines @@
   }
 }
 
 // Test crashing with ASAN on Android. crbug.com/468762
 #if defined(OS_ANDROID) && defined(ADDRESS_SANITIZER)
 #define MAYBE_TestAllSampleRates DISABLED_TestAllSampleRates
 #else
 #define MAYBE_TestAllSampleRates TestAllSampleRates
 #endif
 TEST_F(MediaStreamAudioProcessorTest, MAYBE_TestAllSampleRates) {
-  base::MessageLoop message_loop;
   MockConstraintFactory constraint_factory;
   scoped_refptr<WebRtcAudioDeviceImpl> webrtc_audio_device(
       new WebRtcAudioDeviceImpl());
   scoped_refptr<MediaStreamAudioProcessor> audio_processor(
       new rtc::RefCountedObject<MediaStreamAudioProcessor>(
           constraint_factory.CreateWebMediaConstraints(), input_device_params_,
           webrtc_audio_device.get()));
   EXPECT_TRUE(audio_processor->has_audio_processing());
 
   static const int kSupportedSampleRates[] =
@@ skipping 16 matching lines @@
 
   // Stop |audio_processor| so that it removes itself from
   // |webrtc_audio_device| and clears its pointer to it.
   audio_processor->Stop();
 }
 
 // Test that if we have an AEC dump message filter created, we are getting it
 // correctly in MSAP. Any IPC messages will be deleted since no sender in the
 // filter will be created.
 TEST_F(MediaStreamAudioProcessorTest, GetAecDumpMessageFilter) {
-  base::MessageLoopForUI message_loop;
   scoped_refptr<AecDumpMessageFilter> aec_dump_message_filter_(
-      new AecDumpMessageFilter(message_loop.task_runner(),
-                               message_loop.task_runner()));
+      new AecDumpMessageFilter(base::ThreadTaskRunnerHandle::Get(),
+                               base::ThreadTaskRunnerHandle::Get()));
 
   MockConstraintFactory constraint_factory;
   scoped_refptr<WebRtcAudioDeviceImpl> webrtc_audio_device(
       new WebRtcAudioDeviceImpl());
   scoped_refptr<MediaStreamAudioProcessor> audio_processor(
       new rtc::RefCountedObject<MediaStreamAudioProcessor>(
           constraint_factory.CreateWebMediaConstraints(), input_device_params_,
           webrtc_audio_device.get()));
 
   EXPECT_TRUE(audio_processor->aec_dump_message_filter_.get());
@@ skipping 63 matching lines @@
   audio_processor->Stop();
 }
 
 // Disabled on android clang builds due to crbug.com/470499
 #if defined(__clang__) && defined(OS_ANDROID)
 #define MAYBE_TestWithKeyboardMicChannel DISABLED_TestWithKeyboardMicChannel
 #else
 #define MAYBE_TestWithKeyboardMicChannel TestWithKeyboardMicChannel
 #endif
 TEST_F(MediaStreamAudioProcessorTest, MAYBE_TestWithKeyboardMicChannel) {
-  base::MessageLoop message_loop;
   MockConstraintFactory constraint_factory;
   constraint_factory.basic().googExperimentalNoiseSuppression.setExact(true);
   scoped_refptr<WebRtcAudioDeviceImpl> webrtc_audio_device(
       new WebRtcAudioDeviceImpl());
   scoped_refptr<MediaStreamAudioProcessor> audio_processor(
       new rtc::RefCountedObject<MediaStreamAudioProcessor>(
           constraint_factory.CreateWebMediaConstraints(), input_device_params_,
           webrtc_audio_device.get()));
   EXPECT_TRUE(audio_processor->has_audio_processing());
 
   media::AudioParameters params(media::AudioParameters::AUDIO_PCM_LOW_LATENCY,
                                 media::CHANNEL_LAYOUT_STEREO_AND_KEYBOARD_MIC,
                                 48000, 16, 512);
   audio_processor->OnCaptureFormatChanged(params);
 
   ProcessDataAndVerifyFormat(audio_processor.get(),
                              kAudioProcessingSampleRate,
                              kAudioProcessingNumberOfChannel,
                              kAudioProcessingSampleRate / 100);
 
   // Stop |audio_processor| so that it removes itself from
   // |webrtc_audio_device| and clears its pointer to it.
   audio_processor->Stop();
 }
 
 }  // namespace content