[fix] fix pacer module crash due to multi-thread

src/media/nvcodec/NvDecoder.cpp

@@ -901,7 +901,7 @@ uint8_t *NvDecoder::GetFrame(int64_t *pTimestamp) {
 
 uint8_t *NvDecoder::GetLockedFrame(int64_t *pTimestamp) {
   uint8_t *pFrame;
-  uint64_t timestamp;
+  uint32_t timestamp;
   if (m_nDecodedFrame > 0) {
     std::lock_guard<std::mutex> lock(m_mtxVPFrame);
     m_nDecodedFrame--;
@@ -924,7 +924,7 @@ void NvDecoder::UnlockFrame(uint8_t **pFrame) {
   m_vpFrame.insert(m_vpFrame.end(), &pFrame[0], &pFrame[1]);
 
   // add a dummy entry for timestamp
-  uint64_t timestamp[2] = {0};
+  uint32_t timestamp[2] = {0};
   m_vTimestamp.insert(m_vTimestamp.end(), &timestamp[0], &timestamp[1]);
 }
 #pragma warning(pop)

src/qos/nack_requester.cc

@@ -56,6 +56,15 @@ int NackRequester::OnReceivedPacket(uint16_t seq_num) {
   return OnReceivedPacket(seq_num, false);
 }
 
+void NackRequester::ProcessNacks() {
+  std::vector<uint16_t> nack_batch = GetNackBatch(kTimeOnly);
+  if (!nack_batch.empty()) {
+    // This batch of NACKs is triggered externally; there is no external
+    // initiator who can batch them with other feedback messages.
+    nack_sender_->SendNack(nack_batch, /*buffering_allowed=*/false);
+  }
+}
+
 int NackRequester::OnReceivedPacket(uint16_t seq_num, bool is_recovered) {
   bool is_retransmitted = true;
 

src/qos/nack_requester.h

@@ -48,6 +48,8 @@ class NackRequester {
   int OnReceivedPacket(uint16_t seq_num);
   int OnReceivedPacket(uint16_t seq_num, bool is_recovered);
 
+  void ProcessNacks();
+
  private:
   void ClearUpTo(uint16_t seq_num);
   void UpdateRtt(int64_t rtt_ms);

src/rtp/rtp_packet/rtp_codec.cc

@@ -547,7 +547,7 @@ void RtpCodec::Encode(uint8_t* buffer, uint32_t size,
     }
   } else if (rtp::PAYLOAD_TYPE::AV1 == payload_type_) {
     std::vector<Obu> obus = ParseObus(buffer, size);
-    uint64_t timestamp =
+    uint32_t timestamp =
         std::chrono::duration_cast<std::chrono::microseconds>(
             std::chrono::system_clock::now().time_since_epoch())
             .count();

src/rtp/rtp_packet/rtp_codec.hxx

@@ -40,7 +40,7 @@ class RtpCodec {
   bool marker_ = false;
   uint32_t payload_type_ = 0;
   uint16_t sequence_number_ = 0;
-  uint64_t timestamp_ = 0;
+  uint32_t timestamp_ = 0;
   uint32_t ssrc_ = 0;
   std::vector<uint32_t> csrcs_;
   uint16_t profile_ = 0;

src/rtp/rtp_packet/rtp_header.h

@@ -52,7 +52,7 @@ struct RTPHeader {
   bool marker_ = false;
   uint8_t payload_type_ = 0;
   uint16_t sequence_number_ = 1;
-  uint64_t timestamp_ = 0;
+  uint32_t timestamp_ = 0;
   uint32_t ssrc_ = 0;
   uint32_t csrcs_[kMaxRtpCsrcSize];
   size_t padding_len;

src/rtp/rtp_packet/rtp_packet.h

@@ -204,7 +204,7 @@ class RtpPacket {
   void SetSequenceNumber(uint16_t sequence_number) {
     sequence_number_ = sequence_number;
   }
-  void SetTimestamp(uint64_t timestamp) { timestamp_ = timestamp; }
+  void SetTimestamp(uint32_t timestamp) { timestamp_ = timestamp; }
   void SetSsrc(uint32_t ssrc) { ssrc_ = ssrc; }
   void SetCsrcs(std::vector<uint32_t> &csrcs) { csrcs_ = csrcs; }
   void SetSize(size_t size) { size_ = size; }
@@ -297,7 +297,7 @@ class RtpPacket {
   bool marker_ = false;
   uint8_t payload_type_ = 0;
   uint16_t sequence_number_ = 1;
-  uint64_t timestamp_ = 0;
+  uint32_t timestamp_ = 0;
   uint32_t ssrc_ = 0;
   std::vector<uint32_t> csrcs_;
 

src/rtp/rtp_packet/rtp_packet.x

@@ -208,7 +208,7 @@ class RtpPacket {
   void SetSequenceNumber(uint16_t sequence_number) {
     sequence_number_ = sequence_number;
   }
-  void SetTimestamp(uint64_t timestamp) { timestamp_ = timestamp; }
+  void SetTimestamp(uint32_t timestamp) { timestamp_ = timestamp; }
   void SetSsrc(uint32_t ssrc) { ssrc_ = ssrc; }
   void SetCsrcs(std::vector<uint32_t> &csrcs) { csrcs_ = csrcs; }
 
@@ -453,7 +453,7 @@ class RtpPacket {
   bool marker_ = false;
   uint8_t payload_type_ = 0;
   uint16_t sequence_number_ = 1;
-  uint64_t timestamp_ = 0;
+  uint32_t timestamp_ = 0;
   uint32_t ssrc_ = 0;
   std::vector<uint32_t> csrcs_;
   uint16_t profile_ = 0;

src/rtp/rtp_packetizer/rtp_packetizer_av1.h

@@ -33,7 +33,7 @@ class RtpPacketizerAv1 : public RtpPacketizer {
   bool marker_;
   uint32_t payload_type_;
   uint16_t sequence_number_;
-  uint64_t timestamp_;
+  uint32_t timestamp_;
   uint32_t ssrc_;
   std::vector<uint32_t> csrcs_;
   uint16_t profile_;

src/rtp/rtp_packetizer/rtp_packetizer_generic.cpp

@@ -54,7 +54,7 @@ std::vector<std::unique_ptr<RtpPacket>> RtpPacketizerGeneric::Build(
       payload_size / MAX_NALU_LEN + (last_packet_size ? 1 : 0);
 
   // TODO: use frame timestamp
-  uint64_t timestamp = std::chrono::duration_cast<std::chrono::microseconds>(
+  uint32_t timestamp = std::chrono::duration_cast<std::chrono::microseconds>(
                            std::chrono::system_clock::now().time_since_epoch())
                            .count();
 

src/rtp/rtp_packetizer/rtp_packetizer_generic.h

@@ -36,7 +36,7 @@ class RtpPacketizerGeneric : public RtpPacketizer {
   bool marker_;
   uint32_t payload_type_;
   uint16_t sequence_number_;
-  uint64_t timestamp_;
+  uint32_t timestamp_;
   uint32_t ssrc_;
   std::vector<uint32_t> csrcs_;
   uint16_t profile_;

src/rtp/rtp_packetizer/rtp_packetizer_h264.cpp

@@ -82,7 +82,13 @@ std::vector<std::unique_ptr<RtpPacket>> RtpPacketizerH264::BuildNalu(
   marker_ = 1;
   payload_type_ = rtp::PAYLOAD_TYPE(payload_type_);
   sequence_number_++;
-  timestamp_ = rtp::kMsToRtpTimestamp * rtp_timestamp;
+
+  // TODO: use frame timestamp
+  uint32_t timestamp = std::chrono::duration_cast<std::chrono::microseconds>(
+                           std::chrono::system_clock::now().time_since_epoch())
+                           .count();
+
+  timestamp_ = timestamp;
 
   if (!csrc_count_) {
   }
@@ -149,7 +155,7 @@ std::vector<std::unique_ptr<RtpPacket>> RtpPacketizerH264::BuildFua(
       payload_size / MAX_NALU_LEN + (last_packet_size ? 1 : 0);
 
   // TODO: use frame timestamp
-  uint64_t timestamp = std::chrono::duration_cast<std::chrono::microseconds>(
+  uint32_t timestamp = std::chrono::duration_cast<std::chrono::microseconds>(
                            std::chrono::system_clock::now().time_since_epoch())
                            .count();
 
@@ -228,6 +234,7 @@ std::vector<std::unique_ptr<RtpPacket>> RtpPacketizerH264::BuildFua(
       std::unique_ptr<webrtc::RtpPacketToSend> rtp_packet =
           std::make_unique<webrtc::RtpPacketToSend>();
       rtp_packet->Build(rtp_packet_frame_.data(), rtp_packet_frame_.size());
+      rtp_packet->Build(rtp_packet_frame_.data(), rtp_packet_frame_.size());
       rtp_packets.emplace_back(std::move(rtp_packet));
     } else {
       std::unique_ptr<RtpPacket> rtp_packet = std::make_unique<RtpPacket>();

src/rtp/rtp_packetizer/rtp_packetizer_h264.h

@@ -45,7 +45,7 @@ class RtpPacketizerH264 : public RtpPacketizer {
   bool marker_;
   uint32_t payload_type_;
   uint16_t sequence_number_;
-  uint64_t timestamp_;
+  uint32_t timestamp_;
   uint32_t ssrc_;
   std::vector<uint32_t> csrcs_;
   uint16_t profile_;

src/transport/channel/rtp_video_receiver.cpp

@@ -12,6 +12,7 @@
 #define NV12_BUFFER_SIZE (1280 * 720 * 3 / 2)
 #define RTCP_RR_INTERVAL 1000
 #define MAX_WAIT_TIME_MS 20      // 20ms
+#define NACK_UPDATE_INTERVAL 20  // 20ms
 
 RtpVideoReceiver::RtpVideoReceiver(std::shared_ptr<SystemClock> clock)
     : ssrc_(GenerateUniqueSsrc()),
@@ -148,11 +149,6 @@ void RtpVideoReceiver::InsertRtpPacket(RtpPacket& rtp_packet) {
          file_rtp_recv_);
 #endif
 
-  receive_side_congestion_controller_.OnReceivedPacket(rtp_packet_received,
-                                                       MediaType::VIDEO);
-
-  nack_->OnReceivedPacket(rtp_packet.SequenceNumber());
-
   last_recv_bytes_ = (uint32_t)rtp_packet.PayloadSize();
   total_rtp_payload_recv_ += (uint32_t)rtp_packet.PayloadSize();
   total_rtp_packets_recv_++;
@@ -167,6 +163,13 @@ void RtpVideoReceiver::InsertRtpPacket(RtpPacket& rtp_packet) {
     io_statistics_->UpdateVideoPacketLossCount(rtp_packet.SequenceNumber());
   }
 
+  uint32_t now_ts = static_cast<uint32_t>(
+      std::chrono::duration_cast<std::chrono::milliseconds>(
+          std::chrono::system_clock::now().time_since_epoch())
+          .count());
+
+  CheckIsTimeUpdateNack(now_ts);
+
   // if (CheckIsTimeSendRR()) {
   //   ReceiverReport rtcp_rr;
   //   RtcpReportBlock report;
@@ -208,14 +211,20 @@ void RtpVideoReceiver::InsertRtpPacket(RtpPacket& rtp_packet) {
     RtpPacketH264 rtp_packet_h264;
     if (rtp_packet_h264.Build(rtp_packet.Buffer().data(), rtp_packet.Size())) {
       rtp_packet_h264.GetFrameHeaderInfo();
-      ProcessH264RtpPacket(rtp_packet_h264);
+      bool is_missing_packet = ProcessH264RtpPacket(rtp_packet_h264);
+      if (!is_missing_packet) {
+        receive_side_congestion_controller_.OnReceivedPacket(
+            rtp_packet_received, MediaType::VIDEO);
+        nack_->OnReceivedPacket(rtp_packet.SequenceNumber(), true);
       } else {
-      LOG_ERROR("Invalid h264 rtp packet");
+        nack_->OnReceivedPacket(rtp_packet.SequenceNumber(), false);
+      }
     }
   }
 }
 
-void RtpVideoReceiver::ProcessH264RtpPacket(RtpPacketH264& rtp_packet_h264) {
+bool RtpVideoReceiver::ProcessH264RtpPacket(RtpPacketH264& rtp_packet_h264) {
+  bool is_missing_packet = false;
   if (!fec_enable_) {
     if (rtp::PAYLOAD_TYPE::H264 == rtp_packet_h264.PayloadType()) {
       rtp::NAL_UNIT_TYPE nalu_type = rtp_packet_h264.NalUnitType();
@@ -232,21 +241,31 @@ void RtpVideoReceiver::ProcessH264RtpPacket(RtpPacketH264& rtp_packet_h264) {
       } else if (rtp::NAL_UNIT_TYPE::FU_A == nalu_type) {
         incomplete_h264_frame_list_[rtp_packet_h264.SequenceNumber()] =
             rtp_packet_h264;
-        if (incomplete_h264_frame_list_.find(
-                rtp_packet_h264.SequenceNumber()) ==
-            incomplete_h264_frame_list_.end()) {
-          LOG_ERROR("missing seq {}", rtp_packet_h264.SequenceNumber());
-        }
         if (rtp_packet_h264.FuAEnd()) {
           CheckIsH264FrameCompletedFuaEndReceived(rtp_packet_h264);
         } else {
           auto missing_seqs_iter =
               missing_sequence_numbers_.find(rtp_packet_h264.Timestamp());
+          auto missing_seqs_wait_ts_iter =
+              missing_sequence_numbers_wait_time_.find(
+                  rtp_packet_h264.Timestamp());
           if (missing_seqs_iter != missing_sequence_numbers_.end()) {
+            if (missing_seqs_wait_ts_iter !=
+                missing_sequence_numbers_wait_time_.end()) {
+              if (clock_->CurrentTime().ms() -
+                      missing_seqs_wait_ts_iter->second <=
+                  MAX_WAIT_TIME_MS) {
                 auto missing_seqs = missing_seqs_iter->second;
                 if (missing_seqs.find(rtp_packet_h264.SequenceNumber()) !=
                     missing_seqs.end()) {
                   CheckIsH264FrameCompletedMissSeqReceived(rtp_packet_h264);
+                  is_missing_packet = true;
+                }
+              } else {
+                missing_sequence_numbers_wait_time_.erase(
+                    missing_seqs_wait_ts_iter);
+                missing_sequence_numbers_.erase(missing_seqs_iter);
+              }
             }
           }
         }
@@ -359,6 +378,8 @@ void RtpVideoReceiver::ProcessH264RtpPacket(RtpPacketH264& rtp_packet_h264) {
   //     }
   //   }
   // }
+
+  return is_missing_packet;
 }
 
 void RtpVideoReceiver::ProcessAv1RtpPacket(RtpPacketAv1& rtp_packet_av1) {
@@ -384,7 +405,7 @@ void RtpVideoReceiver::ProcessAv1RtpPacket(RtpPacketAv1& rtp_packet_av1) {
 
 bool RtpVideoReceiver::CheckIsH264FrameCompletedFuaEndReceived(
     RtpPacketH264& rtp_packet_h264) {
-  uint64_t timestamp = rtp_packet_h264.Timestamp();
+  uint32_t timestamp = rtp_packet_h264.Timestamp();
   uint16_t end_seq = rtp_packet_h264.SequenceNumber();
   fua_end_sequence_numbers_[timestamp] = end_seq;
   uint16_t start_seq = 0;
@@ -427,7 +448,7 @@ bool RtpVideoReceiver::CheckIsH264FrameCompletedMissSeqReceived(
     return false;
   }
 
-  uint64_t timestamp = rtp_packet_h264.Timestamp();
+  uint32_t timestamp = rtp_packet_h264.Timestamp();
   uint16_t end_seq = fua_end_sequence_numbers_[timestamp];
   uint16_t start_seq = 0;
   bool has_start = false;
@@ -468,7 +489,7 @@ bool RtpVideoReceiver::CheckIsH264FrameCompletedMissSeqReceived(
 }
 
 bool RtpVideoReceiver::PopCompleteFrame(uint16_t start_seq, uint16_t end_seq,
-                                        uint64_t timestamp) {
+                                        uint32_t timestamp) {
   size_t complete_frame_size = 0;
   int frame_fragment_count = 0;
 
@@ -630,20 +651,24 @@ void RtpVideoReceiver::SendRemb(int64_t bitrate_bps,
   active_remb_module_->SetRemb(bitrate_bps, std::move(ssrcs));
 }
 
-bool RtpVideoReceiver::CheckIsTimeSendRR() {
-  uint32_t now_ts = static_cast<uint32_t>(
-      std::chrono::duration_cast<std::chrono::milliseconds>(
-          std::chrono::system_clock::now().time_since_epoch())
-          .count());
-
-  if (now_ts - last_send_rtcp_rr_packet_ts_ >= RTCP_RR_INTERVAL) {
-    last_send_rtcp_rr_packet_ts_ = now_ts;
+bool RtpVideoReceiver::CheckIsTimeSendRR(uint32_t now) {
+  if (now - last_send_rtcp_rr_packet_ts_ >= RTCP_RR_INTERVAL) {
+    last_send_rtcp_rr_packet_ts_ = now;
     return true;
   } else {
     return false;
   }
 }
 
+void RtpVideoReceiver::CheckIsTimeUpdateNack(uint32_t now) {
+  if (now - last_nack_update_ts_ >= NACK_UPDATE_INTERVAL) {
+    last_send_rtcp_rr_packet_ts_ = now;
+    if (nack_) {
+      nack_->ProcessNacks();
+    }
+  }
+}
+
 bool RtpVideoReceiver::Process() {
   if (!compelete_video_frame_queue_.isEmpty()) {
     std::optional<ReceivedFrame> video_frame =

src/transport/channel/rtp_video_receiver.h

@@ -55,14 +55,15 @@ class RtpVideoReceiver : public ThreadBase,
   bool CheckIsAv1FrameCompleted(RtpPacketAv1& rtp_packet_av1);
 
  private:
-  void ProcessH264RtpPacket(RtpPacketH264& rtp_packet_h264);
+  bool ProcessH264RtpPacket(RtpPacketH264& rtp_packet_h264);
   bool CheckIsH264FrameCompletedFuaEndReceived(RtpPacketH264& rtp_packet_h264);
   bool CheckIsH264FrameCompletedMissSeqReceived(RtpPacketH264& rtp_packet_h264);
   bool PopCompleteFrame(uint16_t start_seq, uint16_t end_seq,
-                        uint64_t timestamp);
+                        uint32_t timestamp);
 
  private:
-  bool CheckIsTimeSendRR();
+  bool CheckIsTimeSendRR(uint32_t now);
+  void CheckIsTimeUpdateNack(uint32_t now);
   int SendRtcpRR(ReceiverReport& rtcp_rr);
 
   void SendCombinedRtcpPacket(
@@ -106,6 +107,7 @@ class RtpVideoReceiver : public ThreadBase,
   uint32_t total_rtp_payload_recv_ = 0;
 
   uint32_t last_send_rtcp_rr_packet_ts_ = 0;
+  uint32_t last_nack_update_ts_ = 0;
   std::function<int(const char*, size_t)> data_send_func_ = nullptr;
 
  private:

src/transport/channel/video_channel_send.cpp

@@ -107,7 +107,7 @@ void VideoChannelSend::Destroy() {
 
 int VideoChannelSend::SendVideo(std::shared_ptr<EncodedFrame> encoded_frame) {
   if (rtp_video_sender_ && rtp_packetizer_) {
-    int64_t rtp_timestamp =
+    int32_t rtp_timestamp =
         delta_ntp_internal_ms_ +
         static_cast<uint32_t>(encoded_frame->CapturedTimestamp() / 1000);
     std::vector<std::unique_ptr<RtpPacket>> rtp_packets =

src/transport/ice_transport_controller.cpp

@@ -53,8 +53,10 @@ void IceTransportController::Create(
   CreateVideoCodec(clock_, video_codec_payload_type, hardware_acceleration);
   CreateAudioCodec();
 
+  task_queue_ = std::make_shared<TaskQueue>();
   controller_ = std::make_unique<CongestionControl>();
-  packet_sender_ = std::make_shared<PacketSenderImp>(ice_agent, webrtc_clock_);
+  packet_sender_ =
+      std::make_shared<PacketSenderImp>(ice_agent, webrtc_clock_, task_queue_);
   packet_sender_->SetPacingRates(DataRate::BitsPerSec(300000),
                                  DataRate::Zero());
   packet_sender_->SetSendBurstInterval(TimeDelta::Millis(40));
@@ -481,9 +483,13 @@ void IceTransportController::OnReceiverReport(
   msg.receive_time = now;
   msg.start_time = last_report_block_time_;
   msg.end_time = now;
+
+  task_queue_->PostTask([this, msg]() mutable {
     if (controller_) {
       PostUpdates(controller_->OnTransportLossReport(msg));
     }
+  });
+
   last_report_block_time_ = now;
 }
 
@@ -492,18 +498,17 @@ void IceTransportController::OnCongestionControlFeedback(
   std::optional<webrtc::TransportPacketsFeedback> feedback_msg =
       transport_feedback_adapter_.ProcessCongestionControlFeedback(
           feedback, Timestamp::Micros(clock_->CurrentTimeUs()));
-  if (feedback_msg) {
-    HandleTransportPacketsFeedback(*feedback_msg);
+  if (feedback_msg.has_value()) {
+    task_queue_->PostTask([this, feedback_msg]() mutable {
+      if (controller_) {
+        PostUpdates(
+            controller_->OnTransportPacketsFeedback(feedback_msg.value()));
       }
-}
-
-void IceTransportController::HandleTransportPacketsFeedback(
-    const webrtc::TransportPacketsFeedback& feedback) {
-  if (controller_)
-    PostUpdates(controller_->OnTransportPacketsFeedback(feedback));
+    });
 
     UpdateCongestedState();
   }
+}
 
 void IceTransportController::OnReceiveNack(
     const std::vector<uint16_t>& nack_sequence_numbers) {
@@ -512,12 +517,6 @@ void IceTransportController::OnReceiveNack(
   }
 }
 
-void IceTransportController::UpdateControllerWithTimeInterval() {
-  ProcessInterval msg;
-  msg.at_time = Timestamp::Millis(webrtc_clock_->TimeInMilliseconds());
-  PostUpdates(controller_->OnProcessInterval(msg));
-}
-
 void IceTransportController::OnSentRtpPacket(
     const webrtc::RtpPacketToSend& packet) {
   webrtc::PacedPacketInfo pacing_info;
@@ -604,8 +603,11 @@ std::optional<bool> IceTransportController::GetCongestedStateUpdate() const {
 }
 
 bool IceTransportController::Process() {
+  task_queue_->PostTask([this]() mutable {
     webrtc::ProcessInterval msg;
     msg.at_time = Timestamp::Millis(webrtc_clock_->TimeInMilliseconds());
     PostUpdates(controller_->OnProcessInterval(msg));
+  });
+
   return true;
 }

src/transport/ice_transport_controller.h

@@ -24,6 +24,7 @@
 #include "packet_sender.h"
 #include "packet_sender_imp.h"
 #include "resolution_adapter.h"
+#include "task_queue.h"
 #include "transport_feedback_adapter.h"
 #include "video_channel_receive.h"
 #include "video_channel_send.h"
@@ -82,10 +83,7 @@ class IceTransportController
   int CreateAudioCodec();
 
  private:
-  void UpdateControllerWithTimeInterval();
   void OnSentRtpPacket(const webrtc::RtpPacketToSend &packet);
-  void HandleTransportPacketsFeedback(
-      const webrtc::TransportPacketsFeedback &feedback);
   void PostUpdates(webrtc::NetworkControlUpdate update);
   void UpdateControlState();
   void UpdateCongestedState();
@@ -121,6 +119,7 @@ class IceTransportController
   webrtc::TransportFeedbackAdapter transport_feedback_adapter_;
   std::unique_ptr<CongestionControl> controller_;
   BitrateProber prober_;
+  std::shared_ptr<TaskQueue> task_queue_;
   webrtc::DataSize congestion_window_size_;
   bool is_congested_ = false;
 

src/transport/packet_sender/packet_sender_imp.cpp

@@ -6,7 +6,8 @@
 const int PacketSenderImp::kNoPacketHoldback = -1;
 
 PacketSenderImp::PacketSenderImp(std::shared_ptr<IceAgent> ice_agent,
-                                 std::shared_ptr<webrtc::Clock> clock)
+                                 std::shared_ptr<webrtc::Clock> clock,
+                                 std::shared_ptr<TaskQueue> task_queue)
     : ice_agent_(ice_agent),
       clock_(clock),
       pacing_controller_(clock.get(), this),
@@ -18,7 +19,8 @@ PacketSenderImp::PacketSenderImp(std::shared_ptr<IceAgent> ice_agent,
       packet_size_(/*alpha=*/0.95),
       include_overhead_(false),
       last_send_time_(webrtc::Timestamp::Millis(0)),
-      last_call_time_(webrtc::Timestamp::Millis(0)) {}
+      last_call_time_(webrtc::Timestamp::Millis(0)),
+      task_queue_(task_queue) {}
 
 PacketSenderImp::~PacketSenderImp() {}
 
@@ -80,7 +82,7 @@ void PacketSenderImp::SetPacingRates(webrtc::DataRate pacing_rate,
 
 void PacketSenderImp::EnqueuePackets(
     std::vector<std::unique_ptr<webrtc::RtpPacketToSend>> packets) {
-  task_queue_.PostTask([this, packets = std::move(packets)]() mutable {
+  task_queue_->PostTask([this, packets = std::move(packets)]() mutable {
     for (auto &packet : packets) {
       size_t packet_size = packet->payload_size() + packet->padding_size();
       if (include_overhead_) {
@@ -94,7 +96,7 @@ void PacketSenderImp::EnqueuePackets(
 }
 
 void PacketSenderImp::RemovePacketsForSsrc(uint32_t ssrc) {
-  task_queue_.PostTask([this, ssrc] {
+  task_queue_->PostTask([this, ssrc] {
     pacing_controller_.RemovePacketsForSsrc(ssrc);
     MaybeProcessPackets(webrtc::Timestamp::MinusInfinity());
   });
@@ -226,7 +228,7 @@ void PacketSenderImp::MaybeProcessPackets(
   if (next_process_time_.IsMinusInfinity() ||
       next_process_time_ > next_send_time) {
     // Prefer low precision if allowed and not probing.
-    task_queue_.PostDelayedTask(
+    task_queue_->PostDelayedTask(
         [this, next_send_time]() { MaybeProcessPackets(next_send_time); },
         time_to_next_process.RoundUpTo(webrtc::TimeDelta::Millis(1)).ms());
     next_process_time_ = next_send_time;
@@ -281,7 +283,6 @@ int PacketSenderImp::EnqueueRtpPacket(
         rtp_packet_to_send->set_packet_type(webrtc::RtpPacketMediaType::kVideo);
         break;
     }
-
     // webrtc::PacedPacketInfo cluster_info;
     // SendPacket(std::move(rtp_packet_to_send), cluster_info);
 

src/transport/packet_sender/packet_sender_imp.h

@@ -31,7 +31,8 @@ class PacketSenderImp : public PacketSender,
   static const int kNoPacketHoldback;
 
   PacketSenderImp(std::shared_ptr<IceAgent> ice_agent,
-                  std::shared_ptr<webrtc::Clock> clock);
+                  std::shared_ptr<webrtc::Clock> clock,
+                  std::shared_ptr<TaskQueue> task_queue);
   ~PacketSenderImp();
 
  public:
@@ -221,7 +222,7 @@ class PacketSenderImp : public PacketSender,
   // Protects against ProcessPackets reentry from packet sent receipts.
   bool processing_packets_ = false;
 
-  TaskQueue task_queue_;
+  std::shared_ptr<TaskQueue> task_queue_;
   int64_t transport_seq_ = 0;
   std::map<int32_t, int16_t> ssrc_seq_;