Uncomment commented-out sequence-checks in call.cc

webrtc/call/call.cc

 /*
  *  Copyright (c) 2013 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 574 matching lines @@
       received_bytes_per_second_counter_.GetStats();
   if (recv_bytes_per_sec.num_samples > kMinRequiredPeriodicSamples) {
     RTC_HISTOGRAM_COUNTS_100000("WebRTC.Call.BitrateReceivedInKbps",
                                 recv_bytes_per_sec.average * 8 / 1000);
     LOG(LS_INFO) << "WebRTC.Call.BitrateReceivedInBps, "
                  << recv_bytes_per_sec.ToStringWithMultiplier(8);
   }
 }
 
 PacketReceiver* Call::Receiver() {
-  // TODO(solenberg): Some test cases in EndToEndTest use this from a different
-  // thread. Re-enable once that is fixed.
-  //  RTC_DCHECK_CALLED_SEQUENTIALLY(&configuration_sequence_checker_);
+  RTC_DCHECK_CALLED_SEQUENTIALLY(&configuration_sequence_checker_);
   return this;
 }
 
 webrtc::AudioSendStream* Call::CreateAudioSendStream(
     const webrtc::AudioSendStream::Config& config) {
   TRACE_EVENT0("webrtc", "Call::CreateAudioSendStream");
   RTC_DCHECK_CALLED_SEQUENTIALLY(&configuration_sequence_checker_);
   event_log_->LogAudioSendStreamConfig(CreateRtcLogStreamConfig(config));
 
   rtc::Optional<RtpState> suspended_rtp_state;
@@ skipping 291 matching lines @@
     // Remove all SSRCs pointing to the FlexfecReceiveStreamImpl to be
     // destroyed.
     receive_side_cc_.GetRemoteBitrateEstimator(UseSendSideBwe(config))
         ->RemoveStream(ssrc);
   }
 
   delete receive_stream;
 }
 
 Call::Stats Call::GetStats() const {
   // TODO(solenberg): Some test cases in EndToEndTest use this from a different

[eladalon, 2017/08/18 14:37:58]

We can take care of this separately, as it requires another change, and I want
to also make sure the TODO is still relevant, meaning that no calls to
GetStats() from any other threads have been added.

   // thread. Re-enable once that is fixed.
   // RTC_DCHECK_CALLED_SEQUENTIALLY(&configuration_sequence_checker_);
   Stats stats;
   // Fetch available send/receive bitrates.
   uint32_t send_bandwidth = 0;
   transport_send_->send_side_cc()->GetBitrateController()->AvailableBandwidth(
       &send_bandwidth);
   std::vector<unsigned int> ssrcs;
   uint32_t recv_bandwidth = 0;
   receive_side_cc_.GetRemoteBitrateEstimator(false)->LatestEstimate(
@@ skipping 447 matching lines @@
     }
   }
   return DELIVERY_UNKNOWN_SSRC;
 }
 
 PacketReceiver::DeliveryStatus Call::DeliverPacket(
     MediaType media_type,
     const uint8_t* packet,
     size_t length,
     const PacketTime& packet_time) {
-  // TODO(solenberg): Tests call this function on a network thread, libjingle
-  // calls on the worker thread. We should move towards always using a network
-  // thread. Then this check can be enabled.
-  // RTC_DCHECK_CALLED_SEQUENTIALLY(&configuration_sequence_checker_);
+  RTC_DCHECK_CALLED_SEQUENTIALLY(&configuration_sequence_checker_);
   if (RtpHeaderParser::IsRtcp(packet, length))
     return DeliverRtcp(media_type, packet, length);
 
   return DeliverRtp(media_type, packet, length, packet_time);
 }
 
 // TODO(brandtr): Update this member function when we support protecting
 // audio packets with FlexFEC.
 void Call::OnRecoveredPacket(const uint8_t* packet, size_t length) {
   ReadLockScoped read_lock(*receive_crit_);
@@ skipping 31 matching lines @@
       (use_send_side_bwe && header.extension.hasTransportSequenceNumber)) {
     receive_side_cc_.OnReceivedPacket(
         packet.arrival_time_ms(), packet.payload_size() + packet.padding_size(),
         header);
   }
 }
 
 }  // namespace internal
 
 }  // namespace webrtc