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);
+  }
 }
 
 // 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);
-}
-
-int8_t ModuleRtpRtcpImpl::SendPayloadType() const {
-  return rtp_sender_.SendPayloadType();
+  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).
+// TODO(nisse): This is currently called also when receiver_only is true. Hence
+// below checks that rtp_sender_ exists. Does that make any sense?
 RTCPSender::FeedbackState ModuleRtpRtcpImpl::GetFeedbackState() {
-  StreamDataCounters rtp_stats;
-  StreamDataCounters rtx_stats;
-  rtp_sender_.GetDataCounters(&rtp_stats, &rtx_stats);
-
   RTCPSender::FeedbackState state;
-  state.send_payload_type = 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;
+  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, 2017/03/13 12:40:09]

separation receiver and sender reveals this is broken:
send_payload_type seems only used by media receiver to send RPSI message.
but payload_type only set when media sender is present and active.
Which mean sending RPSI is broken - even when it is possible to send an rpsi,
incorrect payload type will be used.

Stefan, do you know if we still want to support RPSI?

May it is better for now to remove sending rpsi (and thus
state.send_payload_type) and then restore it correctly if we want to use it.

+    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);
+  rtp_sender_->SetSendingMediaStatus(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);
 }
 
 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;
 }
 
 int32_t ModuleRtpRtcpImpl::SendRTCPSliceLossIndication(uint8_t picture_id) {
   return rtcp_sender_.SendRTCP(GetFeedbackState(), kRtcpSli, 0, 0, picture_id);
 }
 
 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);
 }
 
 int32_t ModuleRtpRtcpImpl::SendRTCPReferencePictureSelection(
     const uint64_t picture_id) {
   return rtcp_sender_.SendRTCP(GetFeedbackState(), kRtcpRpsi, 0, 0, picture_id);
 }
 
 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