Don't allocate any RTPSender object for a receive only RtpRtcp module

webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.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 52 matching lines @@
 
 // Deprecated.
 int32_t RtpRtcp::SetFecParameters(const FecProtectionParams* delta_params,
                                   const FecProtectionParams* key_params) {
   RTC_DCHECK(delta_params);
   RTC_DCHECK(key_params);
   return SetFecParameters(*delta_params, *key_params) ? 0 : -1;
 }
 
 ModuleRtpRtcpImpl::ModuleRtpRtcpImpl(const Configuration& configuration)
-    : rtp_sender_(configuration.audio,
-                  configuration.clock,
-                  configuration.outgoing_transport,
-                  configuration.paced_sender,
-                  configuration.flexfec_sender,
-                  configuration.transport_sequence_number_allocator,
-                  configuration.transport_feedback_callback,
-                  configuration.send_bitrate_observer,
-                  configuration.send_frame_count_observer,
-                  configuration.send_side_delay_observer,
-                  configuration.event_log,
-                  configuration.send_packet_observer,
-                  configuration.retransmission_rate_limiter,
-                  configuration.overhead_observer),
-      rtcp_sender_(configuration.audio,
+    : rtcp_sender_(configuration.audio,
                    configuration.clock,
                    configuration.receive_statistics,
                    configuration.rtcp_packet_type_counter_observer,
                    configuration.event_log,
                    configuration.outgoing_transport),
       rtcp_receiver_(configuration.clock,
                      configuration.receiver_only,
                      configuration.rtcp_packet_type_counter_observer,
                      configuration.bandwidth_callback,
                      configuration.intra_frame_callback,
                      configuration.transport_feedback_callback,
                      configuration.bitrate_allocation_observer,
                      this),
       clock_(configuration.clock),
       audio_(configuration.audio),
       last_process_time_(configuration.clock->TimeInMilliseconds()),
       last_bitrate_process_time_(configuration.clock->TimeInMilliseconds()),
       last_rtt_process_time_(configuration.clock->TimeInMilliseconds()),
       packet_overhead_(28),  // IPV4 UDP.
       nack_last_time_sent_full_(0),
       nack_last_time_sent_full_prev_(0),
       nack_last_seq_number_sent_(0),
       key_frame_req_method_(kKeyFrameReqPliRtcp),
       remote_bitrate_(configuration.remote_bitrate_estimator),
       rtt_stats_(configuration.rtt_stats),
       rtt_ms_(0) {
-  // Make sure rtcp sender use same timestamp offset as rtp sender.
-  rtcp_sender_.SetTimestampOffset(rtp_sender_.TimestampOffset());
+  if (!configuration.receiver_only) {
+    rtp_sender_.reset(new RTPSender(
+        configuration.audio,
+        configuration.clock,
+        configuration.outgoing_transport,
+        configuration.paced_sender,
+        configuration.flexfec_sender,
+        configuration.transport_sequence_number_allocator,
+        configuration.transport_feedback_callback,
+        configuration.send_bitrate_observer,
+        configuration.send_frame_count_observer,
+        configuration.send_side_delay_observer,
+        configuration.event_log,
+        configuration.send_packet_observer,
+        configuration.retransmission_rate_limiter,
+        configuration.overhead_observer));
+    // Make sure rtcp sender use same timestamp offset as rtp sender.
+    rtcp_sender_.SetTimestampOffset(rtp_sender_->TimestampOffset());
 
-  // Set default packet size limit.
-  // TODO(nisse): Kind-of duplicates
-  // webrtc::VideoSendStream::Config::Rtp::kDefaultMaxPacketSize.
-  const size_t kTcpOverIpv4HeaderSize = 40;
-  SetMaxRtpPacketSize(IP_PACKET_SIZE - kTcpOverIpv4HeaderSize);
+    // Set default packet size limit.
+    // TODO(nisse): Kind-of duplicates
+    // webrtc::VideoSendStream::Config::Rtp::kDefaultMaxPacketSize.
+    const size_t kTcpOverIpv4HeaderSize = 40;
+    SetMaxRtpPacketSize(IP_PACKET_SIZE - kTcpOverIpv4HeaderSize);

[danilchap, 2017/03/16 15:41:12]

rtcp_sender also need to know max packet size.

[nisse-webrtc, 2017/03/17 09:29:44]

Done.

+  }
 }
 
 // Returns the number of milliseconds until the module want a worker thread
 // to call Process.
 int64_t ModuleRtpRtcpImpl::TimeUntilNextProcess() {
   const int64_t now = clock_->TimeInMilliseconds();
   const int64_t kRtpRtcpMaxIdleTimeProcessMs = 5;
   return kRtpRtcpMaxIdleTimeProcessMs - (now - last_process_time_);
 }
 
 // Process any pending tasks such as timeouts (non time critical events).
 void ModuleRtpRtcpImpl::Process() {
   const int64_t now = clock_->TimeInMilliseconds();
   last_process_time_ = now;
 
-  const int64_t kRtpRtcpBitrateProcessTimeMs = 10;
-  if (now >= last_bitrate_process_time_ + kRtpRtcpBitrateProcessTimeMs) {
-    rtp_sender_.ProcessBitrate();
-    last_bitrate_process_time_ = now;
+  if (rtp_sender_) {
+    const int64_t kRtpRtcpBitrateProcessTimeMs = 10;
+    if (now >= last_bitrate_process_time_ + kRtpRtcpBitrateProcessTimeMs) {
+      rtp_sender_->ProcessBitrate();
+      last_bitrate_process_time_ = now;
+    }
   }
-
   const int64_t kRtpRtcpRttProcessTimeMs = 1000;
   bool process_rtt = now >= last_rtt_process_time_ + kRtpRtcpRttProcessTimeMs;
   if (rtcp_sender_.Sending()) {
     // Process RTT if we have received a receiver report and we haven't
     // processed RTT for at least |kRtpRtcpRttProcessTimeMs| milliseconds.
     if (rtcp_receiver_.LastReceivedReceiverReport() >
         last_rtt_process_time_ && process_rtt) {
       std::vector<RTCPReportBlock> receive_blocks;
       rtcp_receiver_.StatisticsReceived(&receive_blocks);
       int64_t max_rtt = 0;
@@ skipping 51 matching lines @@
 
   if (rtcp_sender_.TimeToSendRTCPReport())
     rtcp_sender_.SendRTCP(GetFeedbackState(), kRtcpReport);
 
   if (TMMBR() && rtcp_receiver_.UpdateTmmbrTimers()) {
     rtcp_receiver_.NotifyTmmbrUpdated();
   }
 }
 
 void ModuleRtpRtcpImpl::SetRtxSendStatus(int mode) {
-  rtp_sender_.SetRtxStatus(mode);
+  rtp_sender_->SetRtxStatus(mode);
 }
 
 int ModuleRtpRtcpImpl::RtxSendStatus() const {
-  return rtp_sender_.RtxStatus();
+  return rtp_sender_ ? rtp_sender_->RtxStatus() : kRtxOff;
 }
 
 void ModuleRtpRtcpImpl::SetRtxSsrc(uint32_t ssrc) {
-  rtp_sender_.SetRtxSsrc(ssrc);
+  rtp_sender_->SetRtxSsrc(ssrc);
 }
 
 void ModuleRtpRtcpImpl::SetRtxSendPayloadType(int payload_type,
                                               int associated_payload_type) {
-  rtp_sender_.SetRtxPayloadType(payload_type, associated_payload_type);
+  rtp_sender_->SetRtxPayloadType(payload_type, associated_payload_type);
 }
 
 rtc::Optional<uint32_t> ModuleRtpRtcpImpl::FlexfecSsrc() const {
-  return rtp_sender_.FlexfecSsrc();
+  return rtp_sender_->FlexfecSsrc();
 }
 
 int32_t ModuleRtpRtcpImpl::IncomingRtcpPacket(
     const uint8_t* rtcp_packet,
     const size_t length) {
   return rtcp_receiver_.IncomingPacket(rtcp_packet, length) ? 0 : -1;
 }
 
 int32_t ModuleRtpRtcpImpl::RegisterSendPayload(
     const CodecInst& voice_codec) {
-  return rtp_sender_.RegisterPayload(
+  return rtp_sender_->RegisterPayload(
       voice_codec.plname, voice_codec.pltype, voice_codec.plfreq,
       voice_codec.channels, (voice_codec.rate < 0) ? 0 : voice_codec.rate);
 }
 
 int32_t ModuleRtpRtcpImpl::RegisterSendPayload(const VideoCodec& video_codec) {
-  return rtp_sender_.RegisterPayload(video_codec.plName, video_codec.plType,
+  return rtp_sender_->RegisterPayload(video_codec.plName, video_codec.plType,
                                      90000, 0, 0);
 }
 
 void ModuleRtpRtcpImpl::RegisterVideoSendPayload(int payload_type,
                                                  const char* payload_name) {
   RTC_CHECK_EQ(
-      0, rtp_sender_.RegisterPayload(payload_name, payload_type, 90000, 0, 0));
+      0, rtp_sender_->RegisterPayload(payload_name, payload_type, 90000, 0, 0));
 }
 
 int32_t ModuleRtpRtcpImpl::DeRegisterSendPayload(const int8_t payload_type) {
-  return rtp_sender_.DeRegisterSendPayload(payload_type);
+  return rtp_sender_->DeRegisterSendPayload(payload_type);
 }
 
 uint32_t ModuleRtpRtcpImpl::StartTimestamp() const {
-  return rtp_sender_.TimestampOffset();
+  return rtp_sender_->TimestampOffset();
 }
 
 // Configure start timestamp, default is a random number.
 void ModuleRtpRtcpImpl::SetStartTimestamp(const uint32_t timestamp) {
   rtcp_sender_.SetTimestampOffset(timestamp);
-  rtp_sender_.SetTimestampOffset(timestamp);
+  rtp_sender_->SetTimestampOffset(timestamp);
 }
 
 uint16_t ModuleRtpRtcpImpl::SequenceNumber() const {
-  return rtp_sender_.SequenceNumber();
+  return rtp_sender_->SequenceNumber();
 }
 
 // Set SequenceNumber, default is a random number.
 void ModuleRtpRtcpImpl::SetSequenceNumber(const uint16_t seq_num) {
-  rtp_sender_.SetSequenceNumber(seq_num);
+  rtp_sender_->SetSequenceNumber(seq_num);
 }
 
 void ModuleRtpRtcpImpl::SetRtpState(const RtpState& rtp_state) {
-  rtp_sender_.SetRtpState(rtp_state);
+  rtp_sender_->SetRtpState(rtp_state);
   rtcp_sender_.SetTimestampOffset(rtp_state.start_timestamp);
 }
 
 void ModuleRtpRtcpImpl::SetRtxState(const RtpState& rtp_state) {
-  rtp_sender_.SetRtxRtpState(rtp_state);
+  rtp_sender_->SetRtxRtpState(rtp_state);
 }
 
 RtpState ModuleRtpRtcpImpl::GetRtpState() const {
-  return rtp_sender_.GetRtpState();
+  return rtp_sender_->GetRtpState();
 }
 
 RtpState ModuleRtpRtcpImpl::GetRtxState() const {
-  return rtp_sender_.GetRtxRtpState();
+  return rtp_sender_->GetRtxRtpState();
 }
 
 uint32_t ModuleRtpRtcpImpl::SSRC() const {
-  return rtp_sender_.SSRC();
+  return rtcp_sender_.SSRC();
 }
 
 void ModuleRtpRtcpImpl::SetSSRC(const uint32_t ssrc) {
-  rtp_sender_.SetSSRC(ssrc);
+  if (rtp_sender_) {
+    rtp_sender_->SetSSRC(ssrc);
+  }
   rtcp_sender_.SetSSRC(ssrc);
   SetRtcpReceiverSsrcs(ssrc);
 }
 
 void ModuleRtpRtcpImpl::SetCsrcs(const std::vector<uint32_t>& csrcs) {
   rtcp_sender_.SetCsrcs(csrcs);
-  rtp_sender_.SetCsrcs(csrcs);
+  rtp_sender_->SetCsrcs(csrcs);
 }
 
 // TODO(pbos): Handle media and RTX streams separately (separate RTCP
 // feedbacks).
 RTCPSender::FeedbackState ModuleRtpRtcpImpl::GetFeedbackState() {
-  StreamDataCounters rtp_stats;
-  StreamDataCounters rtx_stats;
-  rtp_sender_.GetDataCounters(&rtp_stats, &rtx_stats);
-
   RTCPSender::FeedbackState state;
-  state.send_payload_type = rtp_sender_.SendPayloadType();
-  state.packets_sent = rtp_stats.transmitted.packets +
-                       rtx_stats.transmitted.packets;
-  state.media_bytes_sent = rtp_stats.transmitted.payload_bytes +
-                           rtx_stats.transmitted.payload_bytes;
+  // This is called also when receiver_only is true. Hence below
+  // checks that rtp_sender_ exists.
+  if (rtp_sender_) {
+    StreamDataCounters rtp_stats;
+    StreamDataCounters rtx_stats;
+    rtp_sender_->GetDataCounters(&rtp_stats, &rtx_stats);
+    state.packets_sent = rtp_stats.transmitted.packets +
+                         rtx_stats.transmitted.packets;
+    state.media_bytes_sent = rtp_stats.transmitted.payload_bytes +
+                             rtx_stats.transmitted.payload_bytes;
+    state.send_payload_type = rtp_sender_->SendPayloadType();
+    state.send_bitrate = rtp_sender_->BitrateSent();
+  } else {
+    // The default value for RTPSender::payload_type_ is -1,
+    // use same value if RTPSender is missing.
+    state.send_payload_type = static_cast<uint8_t>(-1);

[danilchap, 2017/03/16 15:41:12]

send_payload_type is now unused (thanks to rpsi removal) - no need to set it.
send_bitrate is set to 0 in the constructor, so else block can be removed.

[nisse-webrtc, 2017/03/17 09:29:44]

Ok, I'm deleting the else clause and setting of send_payload_type. And then I'll
make a separate small cl to delete send_payload_type.

+    state.send_bitrate = 0;
+  }
   state.module = this;
 
   LastReceivedNTP(&state.last_rr_ntp_secs,
                   &state.last_rr_ntp_frac,
                   &state.remote_sr);
 
   state.has_last_xr_rr =
       rtcp_receiver_.LastReceivedXrReferenceTimeInfo(&state.last_xr_rr);
 
-  uint32_t tmp;
-  BitrateSent(&state.send_bitrate, &tmp, &tmp, &tmp);
   return state;
 }
 
+// TODO(nisse): The
+// TestWithNewVideoJitterBuffer/EndToEndTest.RembWithSendSideBwe/* tests call
+// this method for a receive-only stream, which makes little sense. If that is
+// fixed, the rtp_sender_ check can be removed.
 int32_t ModuleRtpRtcpImpl::SetSendingStatus(const bool sending) {
   if (rtcp_sender_.Sending() != sending) {
     // Sends RTCP BYE when going from true to false
     if (rtcp_sender_.SetSendingStatus(GetFeedbackState(), sending) != 0) {
       LOG(LS_WARNING) << "Failed to send RTCP BYE";
     }
-    if (sending) {
+    if (sending && rtp_sender_) {
       // Update Rtcp receiver config, to track Rtx config changes from
       // the SetRtxStatus and SetRtxSsrc methods.
-      SetRtcpReceiverSsrcs(rtp_sender_.SSRC());
+      SetRtcpReceiverSsrcs(rtp_sender_->SSRC());
     }
   }
   return 0;
 }
 
 bool ModuleRtpRtcpImpl::Sending() const {
   return rtcp_sender_.Sending();
 }
 
 void ModuleRtpRtcpImpl::SetSendingMediaStatus(const bool sending) {
-  rtp_sender_.SetSendingMediaStatus(sending);
+  if (rtp_sender_) {
+    rtp_sender_->SetSendingMediaStatus(sending);
+  } else {
+    RTC_DCHECK(!sending);
+  }
 }
 
 bool ModuleRtpRtcpImpl::SendingMedia() const {
-  return rtp_sender_.SendingMedia();
+  return rtp_sender_ ? rtp_sender_->SendingMedia() : false;
 }
 
 bool ModuleRtpRtcpImpl::SendOutgoingData(
     FrameType frame_type,
     int8_t payload_type,
     uint32_t time_stamp,
     int64_t capture_time_ms,
     const uint8_t* payload_data,
     size_t payload_size,
     const RTPFragmentationHeader* fragmentation,
     const RTPVideoHeader* rtp_video_header,
     uint32_t* transport_frame_id_out) {
   rtcp_sender_.SetLastRtpTime(time_stamp, capture_time_ms);
   // Make sure an RTCP report isn't queued behind a key frame.
   if (rtcp_sender_.TimeToSendRTCPReport(kVideoFrameKey == frame_type)) {
       rtcp_sender_.SendRTCP(GetFeedbackState(), kRtcpReport);
   }
-  return rtp_sender_.SendOutgoingData(
+  return rtp_sender_->SendOutgoingData(
       frame_type, payload_type, time_stamp, capture_time_ms, payload_data,
       payload_size, fragmentation, rtp_video_header, transport_frame_id_out);
 }
 
 bool ModuleRtpRtcpImpl::TimeToSendPacket(uint32_t ssrc,
                                          uint16_t sequence_number,
                                          int64_t capture_time_ms,
                                          bool retransmission,
                                          const PacedPacketInfo& pacing_info) {
-  return rtp_sender_.TimeToSendPacket(ssrc, sequence_number, capture_time_ms,
+  return rtp_sender_->TimeToSendPacket(ssrc, sequence_number, capture_time_ms,
                                       retransmission, pacing_info);
 }
 
 size_t ModuleRtpRtcpImpl::TimeToSendPadding(
     size_t bytes,
     const PacedPacketInfo& pacing_info) {
-  return rtp_sender_.TimeToSendPadding(bytes, pacing_info);
+  return rtp_sender_->TimeToSendPadding(bytes, pacing_info);
 }
 
 size_t ModuleRtpRtcpImpl::MaxPayloadSize() const {
-  return rtp_sender_.MaxPayloadSize();
+  return rtp_sender_->MaxPayloadSize();
 }
 
 size_t ModuleRtpRtcpImpl::MaxRtpPacketSize() const {
-  return rtp_sender_.MaxRtpPacketSize();
+  return rtp_sender_->MaxRtpPacketSize();
 }
 
 void ModuleRtpRtcpImpl::SetMaxRtpPacketSize(size_t rtp_packet_size) {
   RTC_DCHECK_LE(rtp_packet_size, IP_PACKET_SIZE)
       << "rtp packet size too large: " << rtp_packet_size;
   RTC_DCHECK_GT(rtp_packet_size, packet_overhead_)
       << "rtp packet size too small: " << rtp_packet_size;
 
   rtcp_sender_.SetMaxRtpPacketSize(rtp_packet_size);
-  rtp_sender_.SetMaxRtpPacketSize(rtp_packet_size);
+  rtp_sender_->SetMaxRtpPacketSize(rtp_packet_size);

[danilchap, 2017/03/16 15:41:13]

since SetMaxRtpPacketSize make sense to call for receiver-only module too,
protect this line

[nisse-webrtc, 2017/03/17 09:29:44]

Done.

 }
 
 RtcpMode ModuleRtpRtcpImpl::RTCP() const {
   return rtcp_sender_.Status();
 }
 
 // Configure RTCP status i.e on/off.
 void ModuleRtpRtcpImpl::SetRTCPStatus(const RtcpMode method) {
   rtcp_sender_.SetRTCPStatus(method);
 }
@@ skipping 80 matching lines @@
 bool ModuleRtpRtcpImpl::RtcpXrRrtrStatus() const {
   return rtcp_sender_.RtcpXrReceiverReferenceTime();
 }
 
 // TODO(asapersson): Replace this method with the one below.
 int32_t ModuleRtpRtcpImpl::DataCountersRTP(
     size_t* bytes_sent,
     uint32_t* packets_sent) const {
   StreamDataCounters rtp_stats;
   StreamDataCounters rtx_stats;
-  rtp_sender_.GetDataCounters(&rtp_stats, &rtx_stats);
+  rtp_sender_->GetDataCounters(&rtp_stats, &rtx_stats);
 
   if (bytes_sent) {
     *bytes_sent = rtp_stats.transmitted.payload_bytes +
                   rtp_stats.transmitted.padding_bytes +
                   rtp_stats.transmitted.header_bytes +
                   rtx_stats.transmitted.payload_bytes +
                   rtx_stats.transmitted.padding_bytes +
                   rtx_stats.transmitted.header_bytes;
   }
   if (packets_sent) {
     *packets_sent = rtp_stats.transmitted.packets +
                     rtx_stats.transmitted.packets;
   }
   return 0;
 }
 
 void ModuleRtpRtcpImpl::GetSendStreamDataCounters(
     StreamDataCounters* rtp_counters,
     StreamDataCounters* rtx_counters) const {
-  rtp_sender_.GetDataCounters(rtp_counters, rtx_counters);
+  rtp_sender_->GetDataCounters(rtp_counters, rtx_counters);
 }
 
 void ModuleRtpRtcpImpl::GetRtpPacketLossStats(
     bool outgoing,
     uint32_t ssrc,
     struct RtpPacketLossStats* loss_stats) const {
   if (!loss_stats) return;
   const PacketLossStats* stats_source = NULL;
   if (outgoing) {
     if (SSRC() == ssrc) {
@@ skipping 34 matching lines @@
 }
 
 void ModuleRtpRtcpImpl::SetREMBData(const uint32_t bitrate,
                                     const std::vector<uint32_t>& ssrcs) {
   rtcp_sender_.SetREMBData(bitrate, ssrcs);
 }
 
 int32_t ModuleRtpRtcpImpl::RegisterSendRtpHeaderExtension(
     const RTPExtensionType type,
     const uint8_t id) {
-  return rtp_sender_.RegisterRtpHeaderExtension(type, id);
+  return rtp_sender_->RegisterRtpHeaderExtension(type, id);
 }
 
 int32_t ModuleRtpRtcpImpl::DeregisterSendRtpHeaderExtension(
     const RTPExtensionType type) {
-  return rtp_sender_.DeregisterRtpHeaderExtension(type);
+  return rtp_sender_->DeregisterRtpHeaderExtension(type);
 }
 
 bool ModuleRtpRtcpImpl::HasBweExtensions() const {
-  return rtp_sender_.IsRtpHeaderExtensionRegistered(
+  return rtp_sender_->IsRtpHeaderExtensionRegistered(
              kRtpExtensionTransportSequenceNumber) ||
-         rtp_sender_.IsRtpHeaderExtensionRegistered(
+         rtp_sender_->IsRtpHeaderExtensionRegistered(
              kRtpExtensionAbsoluteSendTime) ||
-         rtp_sender_.IsRtpHeaderExtensionRegistered(
+         rtp_sender_->IsRtpHeaderExtensionRegistered(
              kRtpExtensionTransmissionTimeOffset);
 }
 
 // (TMMBR) Temporary Max Media Bit Rate.
 bool ModuleRtpRtcpImpl::TMMBR() const {
   return rtcp_sender_.TMMBR();
 }
 
 void ModuleRtpRtcpImpl::SetTMMBRStatus(const bool enable) {
   rtcp_sender_.SetTMMBRStatus(enable);
 }
 
 void ModuleRtpRtcpImpl::SetTmmbn(std::vector<rtcp::TmmbItem> bounding_set) {
   rtcp_sender_.SetTmmbn(std::move(bounding_set));
 }
 
 // Returns the currently configured retransmission mode.
 int ModuleRtpRtcpImpl::SelectiveRetransmissions() const {
-  return rtp_sender_.SelectiveRetransmissions();
+  return rtp_sender_->SelectiveRetransmissions();
 }
 
 // Enable or disable a retransmission mode, which decides which packets will
 // be retransmitted if NACKed.
 int ModuleRtpRtcpImpl::SetSelectiveRetransmissions(uint8_t settings) {
-  return rtp_sender_.SetSelectiveRetransmissions(settings);
+  return rtp_sender_->SetSelectiveRetransmissions(settings);
 }
 
 // Send a Negative acknowledgment packet.
 int32_t ModuleRtpRtcpImpl::SendNACK(const uint16_t* nack_list,
                                     const uint16_t size) {
   for (int i = 0; i < size; ++i) {
     receive_loss_stats_.AddLostPacket(nack_list[i]);
   }
   uint16_t nack_length = size;
   uint16_t start_id = 0;
@@ skipping 50 matching lines @@
   // Send a full NACK list once within every |wait_time|.
   if (rtt_stats_) {
     return now - nack_last_time_sent_full_ > wait_time;
   }
   return now - nack_last_time_sent_full_prev_ > wait_time;
 }
 
 // Store the sent packets, needed to answer to Negative acknowledgment requests.
 void ModuleRtpRtcpImpl::SetStorePacketsStatus(const bool enable,
                                               const uint16_t number_to_store) {
-  rtp_sender_.SetStorePacketsStatus(enable, number_to_store);
+  rtp_sender_->SetStorePacketsStatus(enable, number_to_store);
 }
 
 bool ModuleRtpRtcpImpl::StorePackets() const {
-  return rtp_sender_.StorePackets();
+  return rtp_sender_->StorePackets();
 }
 
 void ModuleRtpRtcpImpl::RegisterRtcpStatisticsCallback(
     RtcpStatisticsCallback* callback) {
   rtcp_receiver_.RegisterRtcpStatisticsCallback(callback);
 }
 
 RtcpStatisticsCallback* ModuleRtpRtcpImpl::GetRtcpStatisticsCallback() {
   return rtcp_receiver_.GetRtcpStatisticsCallback();
 }
 
 bool ModuleRtpRtcpImpl::SendFeedbackPacket(
     const rtcp::TransportFeedback& packet) {
   return rtcp_sender_.SendFeedbackPacket(packet);
 }
 
 // Send a TelephoneEvent tone using RFC 2833 (4733).
 int32_t ModuleRtpRtcpImpl::SendTelephoneEventOutband(
     const uint8_t key,
     const uint16_t time_ms,
     const uint8_t level) {
-  return rtp_sender_.SendTelephoneEvent(key, time_ms, level);
+  return rtp_sender_->SendTelephoneEvent(key, time_ms, level);
 }
 
 int32_t ModuleRtpRtcpImpl::SetAudioPacketSize(
     const uint16_t packet_size_samples) {
   return audio_ ? 0 : -1;
 }
 
 int32_t ModuleRtpRtcpImpl::SetAudioLevel(
     const uint8_t level_d_bov) {
-  return rtp_sender_.SetAudioLevel(level_d_bov);
+  return rtp_sender_->SetAudioLevel(level_d_bov);
 }
 
 int32_t ModuleRtpRtcpImpl::SetKeyFrameRequestMethod(
     const KeyFrameRequestMethod method) {
   key_frame_req_method_ = method;
   return 0;
 }
 
 int32_t ModuleRtpRtcpImpl::RequestKeyFrame() {
   switch (key_frame_req_method_) {
     case kKeyFrameReqPliRtcp:
       return SendRTCP(kRtcpPli);
     case kKeyFrameReqFirRtcp:
       return SendRTCP(kRtcpFir);
   }
   return -1;
 }
 
 void ModuleRtpRtcpImpl::SetUlpfecConfig(int red_payload_type,
                                         int ulpfec_payload_type) {
-  rtp_sender_.SetUlpfecConfig(red_payload_type, ulpfec_payload_type);
+  rtp_sender_->SetUlpfecConfig(red_payload_type, ulpfec_payload_type);
 }
 
 bool ModuleRtpRtcpImpl::SetFecParameters(
     const FecProtectionParams& delta_params,
     const FecProtectionParams& key_params) {
-  return rtp_sender_.SetFecParameters(delta_params, key_params);
+  return rtp_sender_->SetFecParameters(delta_params, key_params);
 }
 
 void ModuleRtpRtcpImpl::SetRemoteSSRC(const uint32_t ssrc) {
   // Inform about the incoming SSRC.
   rtcp_sender_.SetRemoteSSRC(ssrc);
   rtcp_receiver_.SetRemoteSSRC(ssrc);
 }
 
 void ModuleRtpRtcpImpl::BitrateSent(uint32_t* total_rate,
                                     uint32_t* video_rate,
                                     uint32_t* fec_rate,
                                     uint32_t* nack_rate) const {
-  *total_rate = rtp_sender_.BitrateSent();
-  *video_rate = rtp_sender_.VideoBitrateSent();
-  *fec_rate = rtp_sender_.FecOverheadRate();
-  *nack_rate = rtp_sender_.NackOverheadRate();
+  *total_rate = rtp_sender_->BitrateSent();
+  *video_rate = rtp_sender_->VideoBitrateSent();
+  *fec_rate = rtp_sender_->FecOverheadRate();
+  *nack_rate = rtp_sender_->NackOverheadRate();
 }
 
 void ModuleRtpRtcpImpl::OnRequestSendReport() {
   SendRTCP(kRtcpSr);
 }
 
 void ModuleRtpRtcpImpl::OnReceivedNack(
     const std::vector<uint16_t>& nack_sequence_numbers) {
+  if (!rtp_sender_)
+    return;
+
   for (uint16_t nack_sequence_number : nack_sequence_numbers) {
     send_loss_stats_.AddLostPacket(nack_sequence_number);
   }
-  if (!rtp_sender_.StorePackets() ||
+  if (!rtp_sender_->StorePackets() ||
       nack_sequence_numbers.size() == 0) {
     return;
   }
   // Use RTT from RtcpRttStats class if provided.
   int64_t rtt = rtt_ms();
   if (rtt == 0) {
     rtcp_receiver_.RTT(rtcp_receiver_.RemoteSSRC(), NULL, &rtt, NULL, NULL);
   }
-  rtp_sender_.OnReceivedNack(nack_sequence_numbers, rtt);
+  rtp_sender_->OnReceivedNack(nack_sequence_numbers, rtt);
 }
 
 void ModuleRtpRtcpImpl::OnReceivedRtcpReportBlocks(
     const ReportBlockList& report_blocks) {
-  rtp_sender_.OnReceivedRtcpReportBlocks(report_blocks);
+  if (rtp_sender_)
+    rtp_sender_->OnReceivedRtcpReportBlocks(report_blocks);
 }
 
 bool ModuleRtpRtcpImpl::LastReceivedNTP(
     uint32_t* rtcp_arrival_time_secs,  // When we got the last report.
     uint32_t* rtcp_arrival_time_frac,
     uint32_t* remote_sr) const {
   // Remote SR: NTP inside the last received (mid 16 bits from sec and frac).
   uint32_t ntp_secs = 0;
   uint32_t ntp_frac = 0;
 
@@ skipping 17 matching lines @@
 int64_t ModuleRtpRtcpImpl::RtcpReportInterval() {
   if (audio_)
     return RTCP_INTERVAL_AUDIO_MS;
   else
     return RTCP_INTERVAL_VIDEO_MS;
 }
 
 void ModuleRtpRtcpImpl::SetRtcpReceiverSsrcs(uint32_t main_ssrc) {
   std::set<uint32_t> ssrcs;
   ssrcs.insert(main_ssrc);
-  if (rtp_sender_.RtxStatus() != kRtxOff)
-    ssrcs.insert(rtp_sender_.RtxSsrc());
+  if (RtxSendStatus() != kRtxOff)
+    ssrcs.insert(rtp_sender_->RtxSsrc());
   rtcp_receiver_.SetSsrcs(main_ssrc, ssrcs);
 }
 
 void ModuleRtpRtcpImpl::set_rtt_ms(int64_t rtt_ms) {
   rtc::CritScope cs(&critical_section_rtt_);
   rtt_ms_ = rtt_ms;
 }
 
 int64_t ModuleRtpRtcpImpl::rtt_ms() const {
   rtc::CritScope cs(&critical_section_rtt_);
   return rtt_ms_;
 }
 
 void ModuleRtpRtcpImpl::RegisterSendChannelRtpStatisticsCallback(
     StreamDataCountersCallback* callback) {
-  rtp_sender_.RegisterRtpStatisticsCallback(callback);
+  rtp_sender_->RegisterRtpStatisticsCallback(callback);
 }
 
 StreamDataCountersCallback*
     ModuleRtpRtcpImpl::GetSendChannelRtpStatisticsCallback() const {
-  return rtp_sender_.GetRtpStatisticsCallback();
+  return rtp_sender_->GetRtpStatisticsCallback();
 }
 
 void ModuleRtpRtcpImpl::SetVideoBitrateAllocation(
     const BitrateAllocation& bitrate) {
   rtcp_sender_.SetVideoBitrateAllocation(bitrate);
 }
 }  // namespace webrtc