Remove PacketRouter sender distinction.

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 705 matching lines @@
   configuration.bandwidth_callback = rtcp_observer_.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));
+  _rtpRtcpModule->SetSendingMediaStatus(false);
 
   statistics_proxy_.reset(new StatisticsProxy(_rtpRtcpModule->SSRC()));
   rtp_receive_statistics_->RegisterRtcpStatisticsCallback(
       statistics_proxy_.get());
 
   Config audioproc_config;
   audioproc_config.Set<ExperimentalAgc>(new ExperimentalAgc(false));
   rx_audioproc_.reset(AudioProcessing::Create(audioproc_config));
 }
 
@@ skipping 261 matching lines @@
   // Resume the previous sequence number which was reset by StopSend().
   // This needs to be done before |sending| is set to true.
   if (send_sequence_number_)
     SetInitSequenceNumber(send_sequence_number_);
 
   if (channel_state_.Get().sending) {
     return 0;
   }
   channel_state_.SetSending(true);
 
+  _rtpRtcpModule->SetSendingMediaStatus(true);
   if (_rtpRtcpModule->SetSendingStatus(true) != 0) {
     _engineStatisticsPtr->SetLastError(
         VE_RTP_RTCP_MODULE_ERROR, kTraceError,
         "StartSend() RTP/RTCP failed to start sending");
+    _rtpRtcpModule->SetSendingMediaStatus(false);
     rtc::CritScope cs(&_callbackCritSect);
     channel_state_.SetSending(false);
     return -1;
   }
 
   return 0;
 }
 
 int32_t Channel::StopSend() {
   WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId, _channelId),
@@ skipping 11 matching lines @@
   // https://code.google.com/p/webrtc/issues/detail?id=2111 .
   send_sequence_number_ = _rtpRtcpModule->SequenceNumber();
 
   // Reset sending SSRC and sequence number and triggers direct transmission
   // of RTCP BYE
   if (_rtpRtcpModule->SetSendingStatus(false) == -1) {
     _engineStatisticsPtr->SetLastError(
         VE_RTP_RTCP_MODULE_ERROR, kTraceWarning,
         "StartSend() RTP/RTCP failed to stop sending");
   }
+  _rtpRtcpModule->SetSendingMediaStatus(false);
 
   return 0;
 }
 
 int32_t Channel::StartReceiving() {
   WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId, _channelId),
                "Channel::StartReceiving()");
   if (channel_state_.Get().receiving) {
     return 0;
   }
@@ skipping 1408 matching lines @@
     TransportFeedbackObserver* transport_feedback_observer,
     PacketRouter* packet_router) {
   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->AddRtpModule(_rtpRtcpModule.get());
   packet_router_ = packet_router;
 }
 
 void Channel::RegisterReceiverCongestionControlObjects(
     PacketRouter* packet_router) {
   RTC_DCHECK(packet_router && !packet_router_);
-  packet_router->AddRtpModule(_rtpRtcpModule.get(), false);
+  packet_router->AddRtpModule(_rtpRtcpModule.get());
   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_->RemoveRtpModule(_rtpRtcpModule.get());
   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);
 }
 
@@ 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