Eliminate RtpVideoStreamReceiver::receive_cs_ in favor of using a SequencedTaskChecker

webrtc/video/rtp_video_stream_receiver.h

Index: webrtc/video/rtp_video_stream_receiver.h
diff --git a/webrtc/video/rtp_video_stream_receiver.h b/webrtc/video/rtp_video_stream_receiver.h
index 4d63d7648048ad05af15a0f593a03c2710bd7946..15af2e60eec9062dae33e675b5f8b1e8ee41dc5c 100644
--- a/webrtc/video/rtp_video_stream_receiver.h
+++ b/webrtc/video/rtp_video_stream_receiver.h
@@ -32,7 +32,7 @@
 #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 {
@@ -183,11 +183,11 @@ class RtpVideoStreamReceiver : public RtpData,
   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_;
 
@@ -210,10 +210,8 @@ class RtpVideoStreamReceiver : public RtpData,
 
   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