Add thread check to ModuleProcessThread::DeRegisterModule

webrtc/voice_engine/channel.h

 /*
  *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
  *
  *  Use of this source code is governed by a BSD-style license
  *  that can be found in the LICENSE file in the root of the source
  *  tree. An additional intellectual property rights grant can be found
  *  in the file PATENTS.  All contributing project authors may
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
 #ifndef WEBRTC_VOICE_ENGINE_CHANNEL_H_
 #define WEBRTC_VOICE_ENGINE_CHANNEL_H_
 
 #include <memory>
 
 #include "webrtc/api/audio/audio_mixer.h"
 #include "webrtc/api/call/audio_sink.h"
 #include "webrtc/base/criticalsection.h"
 #include "webrtc/base/optional.h"
+#include "webrtc/base/thread_checker.h"
 #include "webrtc/common_audio/resampler/include/push_resampler.h"
 #include "webrtc/common_types.h"
 #include "webrtc/modules/audio_coding/acm2/codec_manager.h"
 #include "webrtc/modules/audio_coding/acm2/rent_a_codec.h"
 #include "webrtc/modules/audio_coding/include/audio_coding_module.h"
 #include "webrtc/modules/audio_conference_mixer/include/audio_conference_mixer_defines.h"
 #include "webrtc/modules/audio_processing/rms_level.h"
 #include "webrtc/modules/rtp_rtcp/include/remote_ntp_time_estimator.h"
 #include "webrtc/modules/rtp_rtcp/include/rtp_header_parser.h"
 #include "webrtc/modules/rtp_rtcp/include/rtp_rtcp.h"
@@ skipping 113 matching lines @@
   virtual ~Channel();
   static int32_t CreateChannel(
       Channel*& channel,
       int32_t channelId,
       uint32_t instanceId,
       const VoEBase::ChannelConfig& config);
   Channel(int32_t channelId,
           uint32_t instanceId,
           const VoEBase::ChannelConfig& config);
   int32_t Init();
+  void Terminate();
   int32_t SetEngineInformation(Statistics& engineStatistics,
                                OutputMixer& outputMixer,
                                ProcessThread& moduleProcessThread,
                                AudioDeviceModule& audioDeviceModule,
                                VoiceEngineObserver* voiceEngineObserver,
                                rtc::CriticalSection* callbackCritSect);
   int32_t UpdateLocalTimeStamp();
 
   void SetSink(std::unique_ptr<AudioSinkInterface> sink);
 
@@ skipping 334 matching lines @@
 
   bool pacing_enabled_;
   PacketRouter* packet_router_ = nullptr;
   std::unique_ptr<TransportFeedbackProxy> feedback_observer_proxy_;
   std::unique_ptr<TransportSequenceNumberProxy> seq_num_allocator_proxy_;
   std::unique_ptr<RtpPacketSenderProxy> rtp_packet_sender_proxy_;
   std::unique_ptr<RateLimiter> retransmission_rate_limiter_;
 
   // TODO(ossu): Remove once GetAudioDecoderFactory() is no longer needed.
   rtc::scoped_refptr<AudioDecoderFactory> decoder_factory_;
+
+  rtc::ThreadChecker construction_thread_;
 };
 
 }  // namespace voe
 }  // namespace webrtc
 
 #endif  // WEBRTC_VOICE_ENGINE_CHANNEL_H_