Eliminate RtpVideoStreamReceiver::receive_cs_ in favor of using a SequencedTaskChecker

webrtc/video/rtp_video_stream_receiver.cc

 /*
  *  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 296 matching lines @@
     const int frequency,
     const size_t channels,
     const uint32_t rate) {
   RTC_NOTREACHED();
   return 0;
 }
 
 // This method handles both regular RTP packets and packets recovered
 // via FlexFEC.
 void RtpVideoStreamReceiver::OnRtpPacket(const RtpPacketReceived& packet) {
-  // TODO(eladalon): https://bugs.chromium.org/p/webrtc/issues/detail?id=8056
-  //  RTC_DCHECK_RUN_ON(&worker_thread_checker_);
+  RTC_DCHECK_CALLED_SEQUENTIALLY(&worker_task_checker_);
 
-  {
-    rtc::CritScope lock(&receive_cs_);
-    if (!receiving_) {
-      return;
-    }
+  if (!receiving_) {
+    return;
+  }
 
-    if (!packet.recovered()) {
-      int64_t now_ms = clock_->TimeInMilliseconds();
+  if (!packet.recovered()) {
+    int64_t now_ms = clock_->TimeInMilliseconds();
 
-      // Periodically log the RTP header of incoming packets.
-      if (now_ms - last_packet_log_ms_ > kPacketLogIntervalMs) {
-        std::stringstream ss;
-        ss << "Packet received on SSRC: " << packet.Ssrc()
-           << " with payload type: " << static_cast<int>(packet.PayloadType())
-           << ", timestamp: " << packet.Timestamp()
-           << ", sequence number: " << packet.SequenceNumber()
-           << ", arrival time: " << packet.arrival_time_ms();
-        int32_t time_offset;
-        if (packet.GetExtension<TransmissionOffset>(&time_offset)) {
-          ss << ", toffset: " << time_offset;
-        }
-        uint32_t send_time;
-        if (packet.GetExtension<AbsoluteSendTime>(&send_time)) {
-          ss << ", abs send time: " << send_time;
-        }
-        LOG(LS_INFO) << ss.str();
-        last_packet_log_ms_ = now_ms;
+    // Periodically log the RTP header of incoming packets.
+    if (now_ms - last_packet_log_ms_ > kPacketLogIntervalMs) {
+      std::stringstream ss;
+      ss << "Packet received on SSRC: " << packet.Ssrc()
+         << " with payload type: " << static_cast<int>(packet.PayloadType())
+         << ", timestamp: " << packet.Timestamp()
+         << ", sequence number: " << packet.SequenceNumber()
+         << ", arrival time: " << packet.arrival_time_ms();
+      int32_t time_offset;
+      if (packet.GetExtension<TransmissionOffset>(&time_offset)) {
+        ss << ", toffset: " << time_offset;
       }
+      uint32_t send_time;
+      if (packet.GetExtension<AbsoluteSendTime>(&send_time)) {
+        ss << ", abs send time: " << send_time;
+      }
+      LOG(LS_INFO) << ss.str();
+      last_packet_log_ms_ = now_ms;
     }
   }
 
   // TODO(nisse): Delete use of GetHeader, but needs refactoring of
   // ReceivePacket and IncomingPacket methods below.
   RTPHeader header;
   packet.GetHeader(&header);
 
   header.payload_type_frequency = kVideoPayloadTypeFrequency;
 
@@ skipping 77 matching lines @@
 rtc::Optional<int64_t> RtpVideoStreamReceiver::LastReceivedPacketMs() const {
   return packet_buffer_->LastReceivedPacketMs();
 }
 
 rtc::Optional<int64_t> RtpVideoStreamReceiver::LastReceivedKeyframePacketMs()
     const {
   return packet_buffer_->LastReceivedKeyframePacketMs();
 }
 
 void RtpVideoStreamReceiver::AddSecondarySink(RtpPacketSinkInterface* sink) {
-  // TODO(eladalon): https://bugs.chromium.org/p/webrtc/issues/detail?id=8056
-  //  RTC_DCHECK_RUN_ON(&worker_thread_checker_);
+  RTC_DCHECK_CALLED_SEQUENTIALLY(&worker_task_checker_);
   RTC_DCHECK(std::find(secondary_sinks_.cbegin(), secondary_sinks_.cend(),
                        sink) == secondary_sinks_.cend());
   secondary_sinks_.push_back(sink);
 }
 
 void RtpVideoStreamReceiver::RemoveSecondarySink(
     const RtpPacketSinkInterface* sink) {
-  // TODO(eladalon): https://bugs.chromium.org/p/webrtc/issues/detail?id=8056
-  //  RTC_DCHECK_RUN_ON(&worker_thread_checker_);
+  RTC_DCHECK_CALLED_SEQUENTIALLY(&worker_task_checker_);
   auto it = std::find(secondary_sinks_.begin(), secondary_sinks_.end(), sink);
   if (it == secondary_sinks_.end()) {
     // We might be rolling-back a call whose setup failed mid-way. In such a
     // case, it's simpler to remove "everything" rather than remember what
     // has already been added.
     LOG(LS_WARNING) << "Removal of unknown sink.";
     return;
   }
   secondary_sinks_.erase(it);
 }
@@ skipping 13 matching lines @@
   if (!rtp_payload_registry_.GetPayloadSpecifics(header.payloadType,
                                                  &payload_specific)) {
     return;
   }
   rtp_receiver_->IncomingRtpPacket(header, payload, payload_length,
                                    payload_specific, in_order);
 }
 
 void RtpVideoStreamReceiver::ParseAndHandleEncapsulatingHeader(
     const uint8_t* packet, size_t packet_length, const RTPHeader& header) {
+  RTC_DCHECK_CALLED_SEQUENTIALLY(&worker_task_checker_);
   if (rtp_payload_registry_.IsRed(header)) {
     int8_t ulpfec_pt = rtp_payload_registry_.ulpfec_payload_type();
     if (packet[header.headerLength] == ulpfec_pt) {
       rtp_receive_statistics_->FecPacketReceived(header, packet_length);
       // Notify video_receiver about received FEC packets to avoid NACKing these
       // packets.
       NotifyReceiverOfFecPacket(header);
     }
     if (ulpfec_receiver_->AddReceivedRedPacket(header, packet, packet_length,
                                                ulpfec_pt) != 0) {
       return;
     }
     ulpfec_receiver_->ProcessReceivedFec();
   } else if (rtp_payload_registry_.IsRtx(header)) {
     if (header.headerLength + header.paddingLength == packet_length) {
       // This is an empty packet and should be silently dropped before trying to
       // parse the RTX header.
       return;
     }
     // Remove the RTX header and parse the original RTP header.
     if (packet_length < header.headerLength)
       return;
     if (packet_length > sizeof(restored_packet_))
       return;
-    rtc::CritScope lock(&receive_cs_);
     if (restored_packet_in_use_) {
       LOG(LS_WARNING) << "Multiple RTX headers detected, dropping packet.";
       return;
     }
     if (!rtp_payload_registry_.RestoreOriginalPacket(
             restored_packet_, packet, &packet_length, config_.rtp.remote_ssrc,
             header)) {
       LOG(LS_WARNING) << "Incoming RTX packet: Invalid RTP header ssrc: "
                       << header.ssrc << " payload type: "
                       << static_cast<int>(header.payloadType);
@@ skipping 37 matching lines @@
   if (header.extension.has_video_timing) {
     rtp_header.type.Video.video_timing = header.extension.video_timing;
   }
   rtp_header.type.Video.playout_delay = header.extension.playout_delay;
 
   OnReceivedPayloadData(nullptr, 0, &rtp_header);
 }
 
 bool RtpVideoStreamReceiver::DeliverRtcp(const uint8_t* rtcp_packet,
                                          size_t rtcp_packet_length) {
-  {
-    rtc::CritScope lock(&receive_cs_);
-    if (!receiving_) {
-      return false;
-    }
+  RTC_DCHECK_CALLED_SEQUENTIALLY(&worker_task_checker_);
+
+  if (!receiving_) {
+    return false;
   }
 
   rtp_rtcp_->IncomingRtcpPacket(rtcp_packet, rtcp_packet_length);
 
   int64_t rtt = 0;
   rtp_rtcp_->RTT(rtp_receiver_->SSRC(), &rtt, nullptr, nullptr, nullptr);
   if (rtt == 0) {
     // Waiting for valid rtt.
     return true;
   }
@@ skipping 41 matching lines @@
     reference_finder_->ClearTo(seq_num);
   }
 }
 
 void RtpVideoStreamReceiver::SignalNetworkState(NetworkState state) {
   rtp_rtcp_->SetRTCPStatus(state == kNetworkUp ? config_.rtp.rtcp_mode
                                                : RtcpMode::kOff);
 }
 
 void RtpVideoStreamReceiver::StartReceive() {
-  rtc::CritScope lock(&receive_cs_);
+  RTC_DCHECK_CALLED_SEQUENTIALLY(&worker_task_checker_);
   receiving_ = true;
 }
 
 void RtpVideoStreamReceiver::StopReceive() {
-  rtc::CritScope lock(&receive_cs_);
+  RTC_DCHECK_CALLED_SEQUENTIALLY(&worker_task_checker_);
   receiving_ = false;
 }
 
 bool RtpVideoStreamReceiver::IsPacketInOrder(const RTPHeader& header) const {
   StreamStatistician* statistician =
       rtp_receive_statistics_->GetStatistician(header.ssrc);
   if (!statistician)
     return false;
   return statistician->IsPacketInOrder(header.sequenceNumber);
 }
@@ skipping 62 matching lines @@
     return;
 
   if (!sprop_decoder.DecodeSprop(sprop_base64_it->second.c_str()))
     return;
 
   tracker_.InsertSpsPpsNalus(sprop_decoder.sps_nalu(),
                              sprop_decoder.pps_nalu());
 }
 
 }  // namespace webrtc