Remove |running_| state from NackModule to avoid running a tight loop.

webrtc/video/rtp_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 168 matching lines @@
   }
 
   if (config_.rtp.rtcp_xr.receiver_reference_time_report)
     rtp_rtcp_->SetRtcpXrRrtrStatus(true);
 
   // Stats callback for CNAME changes.
   rtp_rtcp_->RegisterRtcpStatisticsCallback(receive_stats_proxy);
 
   process_thread_->RegisterModule(rtp_rtcp_.get());
 
-  nack_module_.reset(
-      new NackModule(clock_, nack_sender, keyframe_request_sender));
-  if (config_.rtp.nack.rtp_history_ms == 0)
-    nack_module_->Stop();
-  process_thread_->RegisterModule(nack_module_.get());
+  if (config_.rtp.nack.rtp_history_ms != 0) {
+    nack_module_.reset(
+        new NackModule(clock_, nack_sender, keyframe_request_sender));
+    process_thread_->RegisterModule(nack_module_.get());
+  }
 
   packet_buffer_ = video_coding::PacketBuffer::Create(
       clock_, kPacketBufferStartSize, kPacketBufferMaxSixe, this);
   reference_finder_.reset(new video_coding::RtpFrameReferenceFinder(this));
 }
 
 RtpStreamReceiver::~RtpStreamReceiver() {
+  if (nack_module_) {
+    process_thread_->DeRegisterModule(nack_module_.get());
+  }
+
   process_thread_->DeRegisterModule(rtp_rtcp_.get());
 
-  process_thread_->DeRegisterModule(nack_module_.get());
-
   packet_router_->RemoveRtpModule(rtp_rtcp_.get());
   rtp_rtcp_->SetREMBStatus(false);
   if (config_.rtp.remb) {
     remb_->RemoveReceiveChannel(rtp_rtcp_.get());
   }
   UpdateHistograms();
 }
 
 bool RtpStreamReceiver::AddReceiveCodec(
     const VideoCodec& video_codec,
@@ skipping 24 matching lines @@
 }
 
 int32_t RtpStreamReceiver::OnReceivedPayloadData(
     const uint8_t* payload_data,
     size_t payload_size,
     const WebRtcRTPHeader* rtp_header) {
   WebRtcRTPHeader rtp_header_with_ntp = *rtp_header;
   rtp_header_with_ntp.ntp_time_ms =
       ntp_estimator_.Estimate(rtp_header->header.timestamp);
   VCMPacket packet(payload_data, payload_size, rtp_header_with_ntp);
-  packet.timesNacked = nack_module_->OnReceivedPacket(packet);
+  packet.timesNacked =
+      nack_module_ ? nack_module_->OnReceivedPacket(packet) : -1;
 
   if (packet.codec == kVideoCodecH264) {
     // Only when we start to receive packets will we know what payload type
     // that will be used. When we know the payload type insert the correct
     // sps/pps into the tracker.
     if (packet.payloadType != last_payload_type_) {
       last_payload_type_ = packet.payloadType;
       InsertSpsPpsIntoTracker(packet.payloadType);
     }
 
@@ skipping 138 matching lines @@
   {
     rtc::CritScope lock(&last_seq_num_cs_);
     video_coding::RtpFrameObject* rtp_frame =
         static_cast<video_coding::RtpFrameObject*>(frame.get());
     last_seq_num_for_pic_id_[rtp_frame->picture_id] = rtp_frame->last_seq_num();
   }
   complete_frame_callback_->OnCompleteFrame(std::move(frame));
 }
 
 void RtpStreamReceiver::OnRttUpdate(int64_t avg_rtt_ms, int64_t max_rtt_ms) {
-  nack_module_->UpdateRtt(max_rtt_ms);
+  if (nack_module_)
+    nack_module_->UpdateRtt(max_rtt_ms);
 }
 
 // TODO(nisse): Drop return value.
 bool RtpStreamReceiver::ReceivePacket(const uint8_t* packet,
                                       size_t packet_length,
                                       const RTPHeader& header,
                                       bool in_order) {
   if (rtp_payload_registry_.IsEncapsulated(header)) {
     return ParseAndHandleEncapsulatingHeader(packet, packet_length, header);
   }
@@ skipping 107 matching lines @@
   if (rtp_rtcp_->RemoteNTP(&ntp_secs, &ntp_frac, nullptr, nullptr,
                            &rtp_timestamp) != 0) {
     // Waiting for RTCP.
     return true;
   }
   ntp_estimator_.UpdateRtcpTimestamp(rtt, ntp_secs, ntp_frac, rtp_timestamp);
 
   return true;
 }
 
 void RtpStreamReceiver::FrameContinuous(uint16_t picture_id) {

[philipel, 2017/02/27 09:31:22]

If |nack_module_| is null then we can return immediately.

[tommi, 2017/02/27 09:35:17]

nice - done.

   int seq_num = -1;
   {
     rtc::CritScope lock(&last_seq_num_cs_);
     auto seq_num_it = last_seq_num_for_pic_id_.find(picture_id);
     if (seq_num_it != last_seq_num_for_pic_id_.end())
       seq_num = seq_num_it->second;
   }
-  if (seq_num != -1)
+  if (seq_num != -1 && nack_module_)
     nack_module_->ClearUpTo(seq_num);
 }
 
 void RtpStreamReceiver::FrameDecoded(uint16_t picture_id) {
   int seq_num = -1;
   {
     rtc::CritScope lock(&last_seq_num_cs_);
     auto seq_num_it = last_seq_num_for_pic_id_.find(picture_id);
     if (seq_num_it != last_seq_num_for_pic_id_.end()) {
       seq_num = seq_num_it->second;
@@ skipping 94 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