Eliminate RtpVideoStreamReceiver::receive_cs_ in favor of using a SequencedTaskChecker

webrtc/video/rtp_video_stream_receiver.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.
  */
 
@@ skipping 14 matching lines @@
 #include "webrtc/modules/rtp_rtcp/include/rtp_payload_registry.h"
 #include "webrtc/modules/rtp_rtcp/include/rtp_rtcp.h"
 #include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
 #include "webrtc/modules/video_coding/h264_sps_pps_tracker.h"
 #include "webrtc/modules/video_coding/include/video_coding_defines.h"
 #include "webrtc/modules/video_coding/packet_buffer.h"
 #include "webrtc/modules/video_coding/rtp_frame_reference_finder.h"
 #include "webrtc/modules/video_coding/sequence_number_util.h"
 #include "webrtc/rtc_base/constructormagic.h"
 #include "webrtc/rtc_base/criticalsection.h"
-#include "webrtc/rtc_base/thread_checker.h"
+#include "webrtc/rtc_base/sequenced_task_checker.h"
 #include "webrtc/typedefs.h"
 
 namespace webrtc {
 
 class NackModule;
 class PacedSender;
 class PacketRouter;
 class ProcessThread;
 class ReceiveStatistics;
 class ReceiveStatisticsProxy;
@@ skipping 130 matching lines @@
   ProcessThread* const process_thread_;
 
   RemoteNtpTimeEstimator ntp_estimator_;
   RTPPayloadRegistry rtp_payload_registry_;
 
   const std::unique_ptr<RtpHeaderParser> rtp_header_parser_;
   const std::unique_ptr<RtpReceiver> rtp_receiver_;
   const std::unique_ptr<ReceiveStatistics> rtp_receive_statistics_;
   std::unique_ptr<UlpfecReceiver> ulpfec_receiver_;
 
-  rtc::CriticalSection receive_cs_;
-  bool receiving_ GUARDED_BY(receive_cs_);
-  uint8_t restored_packet_[IP_PACKET_SIZE] GUARDED_BY(receive_cs_);
-  bool restored_packet_in_use_ GUARDED_BY(receive_cs_);
-  int64_t last_packet_log_ms_ GUARDED_BY(receive_cs_);
+  rtc::SequencedTaskChecker worker_task_checker_;
+  bool receiving_ GUARDED_BY(worker_task_checker_);
+  uint8_t restored_packet_[IP_PACKET_SIZE] GUARDED_BY(worker_task_checker_);
+  bool restored_packet_in_use_ GUARDED_BY(worker_task_checker_);
+  int64_t last_packet_log_ms_ GUARDED_BY(worker_task_checker_);
 
   const std::unique_ptr<RtpRtcp> rtp_rtcp_;
 
   // Members for the new jitter buffer experiment.
   video_coding::OnCompleteFrameCallback* complete_frame_callback_;
   KeyFrameRequestSender* keyframe_request_sender_;
   VCMTiming* timing_;
   std::unique_ptr<NackModule> nack_module_;
   rtc::scoped_refptr<video_coding::PacketBuffer> packet_buffer_;
   std::unique_ptr<video_coding::RtpFrameReferenceFinder> reference_finder_;
   rtc::CriticalSection last_seq_num_cs_;
   std::map<uint16_t, uint16_t, DescendingSeqNumComp<uint16_t>>
       last_seq_num_for_pic_id_ GUARDED_BY(last_seq_num_cs_);
   video_coding::H264SpsPpsTracker tracker_;
   // TODO(johan): Remove pt_codec_params_ once
   // https://bugs.chromium.org/p/webrtc/issues/detail?id=6883 is resolved.
   // Maps a payload type to a map of out-of-band supplied codec parameters.
   std::map<uint8_t, std::map<std::string, std::string>> pt_codec_params_;
   int16_t last_payload_type_ = -1;
 
   bool has_received_frame_;
 
-  // TODO(eladalon): https://bugs.chromium.org/p/webrtc/issues/detail?id=8056
-  //  rtc::ThreadChecker worker_thread_checker_;
-  std::vector<RtpPacketSinkInterface*> secondary_sinks_;  // This needs
-  // to be GUARDED_BY(worker_thread_checker_).
+  std::vector<RtpPacketSinkInterface*> secondary_sinks_
+      GUARDED_BY(worker_task_checker_);
 };
 
 }  // namespace webrtc
 
 #endif  // WEBRTC_VIDEO_RTP_VIDEO_STREAM_RECEIVER_H_