Use separate rtp module lists for send and receive in PacketRouter.

webrtc/voice_engine/channel.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 106 matching lines @@
 class RtpPacketSenderProxy : public RtpPacketSender {
  public:
   RtpPacketSenderProxy() : rtp_packet_sender_(nullptr) {}
 
   void SetPacketSender(RtpPacketSender* rtp_packet_sender) {
     RTC_DCHECK(thread_checker_.CalledOnValidThread());
     rtc::CritScope lock(&crit_);
     rtp_packet_sender_ = rtp_packet_sender;
   }
 
+  bool HasPacketSender() const {
+    RTC_DCHECK(thread_checker_.CalledOnValidThread());
+    rtc::CritScope lock(&crit_);
+    return rtp_packet_sender_ != nullptr;
+  }
+
   // Implements RtpPacketSender.
   void InsertPacket(Priority priority,
                     uint32_t ssrc,
                     uint16_t sequence_number,
                     int64_t capture_time_ms,
                     size_t bytes,
                     bool retransmission) override {
     rtc::CritScope lock(&crit_);
     if (rtp_packet_sender_) {
       rtp_packet_sender_->InsertPacket(priority, ssrc, sequence_number,
@@ skipping 669 matching lines @@
       _previousTimestamp(0),
       _recPacketDelayMs(20),
       _RxVadDetection(false),
       _rxAgcIsEnabled(false),
       _rxNsIsEnabled(false),
       restored_packet_in_use_(false),
       rtcp_observer_(new VoERtcpObserver(this)),
       network_predictor_(new NetworkPredictor(Clock::GetRealTimeClock())),
       associate_send_channel_(ChannelOwner(nullptr)),
       pacing_enabled_(config.Get<VoicePacing>().enabled),
-      feedback_observer_proxy_(pacing_enabled_ ? new TransportFeedbackProxy()
-                                               : nullptr),
-      seq_num_allocator_proxy_(
-          pacing_enabled_ ? new TransportSequenceNumberProxy() : nullptr),
-      rtp_packet_sender_proxy_(pacing_enabled_ ? new RtpPacketSenderProxy()
-                                               : nullptr) {
+      feedback_observer_proxy_(new TransportFeedbackProxy()),
+      seq_num_allocator_proxy_(new TransportSequenceNumberProxy()),
+      rtp_packet_sender_proxy_(new RtpPacketSenderProxy()) {
   WEBRTC_TRACE(kTraceMemory, kTraceVoice, VoEId(_instanceId, _channelId),
                "Channel::Channel() - ctor");
   AudioCodingModule::Config acm_config;
   acm_config.id = VoEModuleId(instanceId, channelId);
   if (config.Get<NetEqCapacityConfig>().enabled) {
     // Clamping the buffer capacity at 20 packets. While going lower will
     // probably work, it makes little sense.
     acm_config.neteq_config.max_packets_in_buffer =
         std::max(20, config.Get<NetEqCapacityConfig>().capacity);
   }
   acm_config.neteq_config.enable_fast_accelerate =
       config.Get<NetEqFastAccelerate>().enabled;
   audio_coding_.reset(AudioCodingModule::Create(acm_config));
 
   _inbandDtmfQueue.ResetDtmf();
   _inbandDtmfGenerator.Init();
   _outputAudioLevel.Clear();
 
   RtpRtcp::Configuration configuration;
   configuration.audio = true;
   configuration.outgoing_transport = this;
   configuration.audio_messages = this;
   configuration.receive_statistics = rtp_receive_statistics_.get();
   configuration.bandwidth_callback = rtcp_observer_.get();
-  configuration.paced_sender = rtp_packet_sender_proxy_.get();
-  configuration.transport_sequence_number_allocator =
-      seq_num_allocator_proxy_.get();
-  configuration.transport_feedback_callback = feedback_observer_proxy_.get();
+  if (pacing_enabled_) {
+    configuration.paced_sender = rtp_packet_sender_proxy_.get();
+    configuration.transport_sequence_number_allocator =
+        seq_num_allocator_proxy_.get();
+    configuration.transport_feedback_callback = feedback_observer_proxy_.get();
+  }
   configuration.event_log = event_log;
 
   _rtpRtcpModule.reset(RtpRtcp::CreateRtpRtcp(configuration));
 
   statistics_proxy_.reset(new StatisticsProxy(_rtpRtcpModule->SSRC()));
   rtp_receive_statistics_->RegisterRtcpStatisticsCallback(
       statistics_proxy_.get());
 
   Config audioproc_config;
   audioproc_config.Set<ExperimentalAgc>(new ExperimentalAgc(false));
@@ skipping 1715 matching lines @@
 }
 
 void Channel::EnableReceiveTransportSequenceNumber(int id) {
   rtp_header_parser_->DeregisterRtpHeaderExtension(
       kRtpExtensionTransportSequenceNumber);
   bool ret = rtp_header_parser_->RegisterRtpHeaderExtension(
       kRtpExtensionTransportSequenceNumber, id);
   RTC_DCHECK(ret);
 }
 
-void Channel::SetCongestionControlObjects(
+void Channel::RegisterSenderCongestionControlObjects(
     RtpPacketSender* rtp_packet_sender,
     TransportFeedbackObserver* transport_feedback_observer,
     PacketRouter* packet_router) {
-  RTC_DCHECK(packet_router != nullptr || packet_router_ != nullptr);
-  if (transport_feedback_observer) {
-    RTC_DCHECK(feedback_observer_proxy_.get());
-    feedback_observer_proxy_->SetTransportFeedbackObserver(
-        transport_feedback_observer);
-  }
-  if (rtp_packet_sender) {
-    RTC_DCHECK(rtp_packet_sender_proxy_.get());
-    rtp_packet_sender_proxy_->SetPacketSender(rtp_packet_sender);
-  }
-  if (seq_num_allocator_proxy_.get()) {
-    seq_num_allocator_proxy_->SetSequenceNumberAllocator(packet_router);
-  }
-  _rtpRtcpModule->SetStorePacketsStatus(rtp_packet_sender != nullptr, 600);
-  if (packet_router != nullptr) {
-    packet_router->AddRtpModule(_rtpRtcpModule.get());
-  } else {
-    packet_router_->RemoveRtpModule(_rtpRtcpModule.get());
-  }
+  RTC_DCHECK(rtp_packet_sender);
+  RTC_DCHECK(transport_feedback_observer);
+  RTC_DCHECK(packet_router && !packet_router_);
+  feedback_observer_proxy_->SetTransportFeedbackObserver(
+      transport_feedback_observer);
+  seq_num_allocator_proxy_->SetSequenceNumberAllocator(packet_router);
+  rtp_packet_sender_proxy_->SetPacketSender(rtp_packet_sender);
+  _rtpRtcpModule->SetStorePacketsStatus(true, 600);
+  packet_router->AddRtpModule(_rtpRtcpModule.get(), true);
   packet_router_ = packet_router;
 }
 
+void Channel::RegisterReceiverCongestionControlObjects(
+    PacketRouter* packet_router) {
+  RTC_DCHECK(packet_router && !packet_router_);
+  packet_router->AddRtpModule(_rtpRtcpModule.get(), false);
+  packet_router_ = packet_router;
+}
+
+void Channel::ResetCongestionControlObjects() {
+  RTC_DCHECK(packet_router_);
+  _rtpRtcpModule->SetStorePacketsStatus(false, 600);
+  feedback_observer_proxy_->SetTransportFeedbackObserver(nullptr);
+  seq_num_allocator_proxy_->SetSequenceNumberAllocator(nullptr);
+  const bool sender = rtp_packet_sender_proxy_->HasPacketSender();
+  packet_router_->RemoveRtpModule(_rtpRtcpModule.get(), sender);
+  packet_router_ = nullptr;
+  rtp_packet_sender_proxy_->SetPacketSender(nullptr);
+}
+
 void Channel::SetRTCPStatus(bool enable) {
   WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId, _channelId),
                "Channel::SetRTCPStatus()");
   _rtpRtcpModule->SetRTCPStatus(enable ? RtcpMode::kCompound : RtcpMode::kOff);
 }
 
 int Channel::GetRTCPStatus(bool& enabled) {
   RtcpMode method = _rtpRtcpModule->RTCP();
   enabled = (method != RtcpMode::kOff);
   return 0;
@@ skipping 1020 matching lines @@
   int64_t min_rtt = 0;
   if (_rtpRtcpModule->RTT(remoteSSRC, &rtt, &avg_rtt, &min_rtt, &max_rtt) !=
       0) {
     return 0;
   }
   return rtt;
 }
 
 }  // namespace voe
 }  // namespace webrtc