Wire up new PacedSender code path.

This CL makes the new code path for paced sending functionally complete.
By default, the field trial WebRTC-Pacer-ReferencePackets is Enabled,
meaning that there is no behavior change unless the field trial is
forced to Disabled. This is done in tests, and can be done on the
command line for manual testing.

Bug: webrtc:10633
Change-Id: I0d66c94ef83b5847dee437a785018f09ba3f828d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/144050
Commit-Queue: Erik Språng <sprang@webrtc.org>
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28497}

modules/pacing/BUILD.gn

@@ -28,6 +28,7 @@ rtc_static_library("pacing") {
   deps = [
     ":interval_budget",
     "..:module_api",
+    "../../api:function_view",
     "../../api/transport:field_trial_based_config",
     "../../api/transport:network_control",
     "../../api/transport:webrtc_key_value_config",

modules/pacing/paced_sender.cc

@@ -12,6 +12,7 @@
 
 #include <algorithm>
 #include <utility>
+#include <vector>
 
 #include "absl/memory/memory.h"
 #include "logging/rtc_event_log/rtc_event_log.h"
@@ -97,7 +98,9 @@ PacedSender::PacedSender(Clock* clock,
       packets_(clock->TimeInMicroseconds()),
       packet_counter_(0),
       queue_time_limit(kMaxQueueLengthMs),
-      account_for_audio_(false) {
+      account_for_audio_(false),
+      legacy_packet_referencing_(
+          !IsDisabled(*field_trials_, "WebRTC-Pacer-LegacyPacketReferencing")) {
   if (!drain_large_queues_) {
     RTC_LOG(LS_WARNING) << "Pacer queues will not be drained,"
                            "pushback experiment must be enabled.";
@@ -328,10 +331,21 @@ void PacedSender::Process() {
   int64_t now_us = clock_->TimeInMicroseconds();
   int64_t elapsed_time_ms = UpdateTimeAndGetElapsedMs(now_us);
   if (ShouldSendKeepalive(now_us)) {
-    critsect_.Leave();
+    if (legacy_packet_referencing_) {
-    size_t bytes_sent = packet_router_->TimeToSendPadding(1, PacedPacketInfo());
+      critsect_.Leave();
-    critsect_.Enter();
+      size_t bytes_sent =
-    OnPaddingSent(bytes_sent);
+          packet_router_->TimeToSendPadding(1, PacedPacketInfo());
+      critsect_.Enter();
+      OnPaddingSent(bytes_sent);
+    } else {
+      critsect_.Leave();
+      std::vector<std::unique_ptr<RtpPacketToSend>> keepalive_packets =
+          packet_router_->GeneratePadding(1);
+      critsect_.Enter();
+      for (auto& packet : keepalive_packets) {
+        EnqueuePacket(std::move(packet));
+      }
+    }
   }
 
   if (paused_)
@@ -364,35 +378,60 @@ void PacedSender::Process() {
 
   bool is_probing = prober_.IsProbing();
   PacedPacketInfo pacing_info;
-  size_t bytes_sent = 0;
+  absl::optional<size_t> recommended_probe_size;
-  size_t recommended_probe_size = 0;
   if (is_probing) {
     pacing_info = prober_.CurrentCluster();
     recommended_probe_size = prober_.RecommendedMinProbeSize();
   }
+
+  size_t bytes_sent = 0;
   // The paused state is checked in the loop since it leaves the critical
   // section allowing the paused state to be changed from other code.
-  while (!packets_.Empty() && !paused_) {
+  while (!paused_) {
     auto* packet = GetPendingPacket(pacing_info);
-    if (packet == nullptr)
+    if (packet == nullptr) {
+      // No packet available to send, check if we should send padding.
+      if (!legacy_packet_referencing_) {
+        size_t padding_bytes_to_add =
+            PaddingBytesToAdd(recommended_probe_size, bytes_sent);
+        if (padding_bytes_to_add > 0) {
+          critsect_.Leave();
+          std::vector<std::unique_ptr<RtpPacketToSend>> padding_packets =
+              packet_router_->GeneratePadding(padding_bytes_to_add);
+          critsect_.Enter();
+          if (padding_packets.empty()) {
+            // No padding packets were generated, quite send loop.
+            break;
+          }
+          for (auto& packet : padding_packets) {
+            EnqueuePacket(std::move(packet));
+          }
+          // Continue loop to send the padding that was just added.
+          continue;
+        }
+      }
+
+      // Can't fetch new packet and no padding to send, exit send loop.
       break;
+    }
 
     std::unique_ptr<RtpPacketToSend> rtp_packet = packet->ReleasePacket();
     const bool owned_rtp_packet = rtp_packet != nullptr;
-
-    critsect_.Leave();
-
     RtpPacketSendResult success;
+
     if (rtp_packet != nullptr) {
+      critsect_.Leave();
       packet_router_->SendPacket(std::move(rtp_packet), pacing_info);
+      critsect_.Enter();
       success = RtpPacketSendResult::kSuccess;
     } else {
+      critsect_.Leave();
       success = packet_router_->TimeToSendPacket(
           packet->ssrc(), packet->sequence_number(), packet->capture_time_ms(),
           packet->is_retransmission(), pacing_info);
+      critsect_.Enter();
     }
 
-    critsect_.Enter();
     if (success == RtpPacketSendResult::kSuccess ||
         success == RtpPacketSendResult::kPacketNotFound) {
       // Packet sent or invalid packet, remove it from queue.
@@ -400,7 +439,7 @@ void PacedSender::Process() {
       bytes_sent += packet->size_in_bytes();
       // Send succeeded, remove it from the queue.
       OnPacketSent(packet);
-      if (is_probing && bytes_sent > recommended_probe_size)
+      if (recommended_probe_size && bytes_sent > *recommended_probe_size)
         break;
     } else if (owned_rtp_packet) {
       // Send failed, but we can't put it back in the queue, remove it without
@@ -414,16 +453,17 @@ void PacedSender::Process() {
     }
   }
 
-  if (packets_.Empty() && !Congested()) {
+  if (legacy_packet_referencing_ && packets_.Empty() && !Congested()) {
     // We can not send padding unless a normal packet has first been sent. If we
     // do, timestamps get messed up.
     if (packet_counter_ > 0) {
-      int padding_needed =
+      int padding_needed = static_cast<int>(
-          static_cast<int>(is_probing ? (recommended_probe_size - bytes_sent)
+          recommended_probe_size ? (*recommended_probe_size - bytes_sent)
-                                      : padding_budget_.bytes_remaining());
+                                 : padding_budget_.bytes_remaining());
       if (padding_needed > 0) {
+        size_t padding_sent = 0;
         critsect_.Leave();
-        size_t padding_sent =
+        padding_sent =
             packet_router_->TimeToSendPadding(padding_needed, pacing_info);
         critsect_.Enter();
         bytes_sent += padding_sent;
@@ -431,6 +471,7 @@ void PacedSender::Process() {
       }
     }
   }
+
   if (is_probing) {
     probing_send_failure_ = bytes_sent == 0;
     if (!probing_send_failure_)
@@ -444,8 +485,41 @@ void PacedSender::ProcessThreadAttached(ProcessThread* process_thread) {
   process_thread_ = process_thread;
 }
 
+size_t PacedSender::PaddingBytesToAdd(
+    absl::optional<size_t> recommended_probe_size,
+    size_t bytes_sent) {
+  if (!packets_.Empty()) {
+    // Actual payload available, no need to add padding.
+    return 0;
+  }
+
+  if (Congested()) {
+    // Don't add padding if congested, even if requested for probing.
+    return 0;
+  }
+
+  if (packet_counter_ == 0) {
+    // We can not send padding unless a normal packet has first been sent. If we
+    // do, timestamps get messed up.
+    return 0;
+  }
+
+  if (recommended_probe_size) {
+    if (*recommended_probe_size > bytes_sent) {
+      return *recommended_probe_size - bytes_sent;
+    }
+    return 0;
+  }
+
+  return padding_budget_.bytes_remaining();
+}
+
 RoundRobinPacketQueue::QueuedPacket* PacedSender::GetPendingPacket(
     const PacedPacketInfo& pacing_info) {
+  if (packets_.Empty()) {
+    return nullptr;
+  }
+
   // Since we need to release the lock in order to send, we first pop the
   // element from the priority queue but keep it in storage, so that we can
   // reinsert it if send fails.

modules/pacing/paced_sender.h

@@ -17,6 +17,7 @@
 #include <memory>
 
 #include "absl/types/optional.h"
+#include "api/function_view.h"
 #include "api/transport/field_trial_based_config.h"
 #include "api/transport/network_types.h"
 #include "api/transport/webrtc_key_value_config.h"
@@ -135,6 +136,10 @@ class PacedSender : public Module, public RtpPacketPacer {
   void UpdateBudgetWithBytesSent(size_t bytes)
       RTC_EXCLUSIVE_LOCKS_REQUIRED(critsect_);
 
+  size_t PaddingBytesToAdd(absl::optional<size_t> recommended_probe_size,
+                           size_t bytes_sent)
+      RTC_EXCLUSIVE_LOCKS_REQUIRED(critsect_);
+
   RoundRobinPacketQueue::QueuedPacket* GetPendingPacket(
       const PacedPacketInfo& pacing_info)
       RTC_EXCLUSIVE_LOCKS_REQUIRED(critsect_);
@@ -195,6 +200,11 @@ class PacedSender : public Module, public RtpPacketPacer {
 
   int64_t queue_time_limit RTC_GUARDED_BY(critsect_);
   bool account_for_audio_ RTC_GUARDED_BY(critsect_);
+
+  // If true, PacedSender should only reference packets as in legacy mode.
+  // If false, PacedSender may have direct ownership of RtpPacketToSend objects.
+  // Defaults to true, will be changed to default false soon.
+  const bool legacy_packet_referencing_;
 };
 }  // namespace webrtc
 #endif  // MODULES_PACING_PACED_SENDER_H_

modules/pacing/packet_router.cc

@@ -146,6 +146,12 @@ RtpRtcp* PacketRouter::FindRtpModule(uint32_t ssrc) {
 void PacketRouter::SendPacket(std::unique_ptr<RtpPacketToSend> packet,
                               const PacedPacketInfo& cluster_info) {
   rtc::CritScope cs(&modules_crit_);
+  // With the new pacer code path, transport sequence numbers are only set here,
+  // on the pacer thread. Therefore we don't need atomics/synchronization.
+  if (packet->IsExtensionReserved<TransportSequenceNumber>() &&
+      packet->SetExtension<TransportSequenceNumber>(transport_seq_)) {
+    ++transport_seq_;
+  }
   for (auto* rtp_module : rtp_send_modules_) {
     if (rtp_module->TrySendPacket(packet.get(), cluster_info)) {
       const bool can_send_padding =
@@ -200,7 +206,8 @@ size_t PacketRouter::TimeToSendPadding(size_t bytes_to_send,
   return total_bytes_sent;
 }
 
-void PacketRouter::GeneratePadding(size_t target_size_bytes) {
+std::vector<std::unique_ptr<RtpPacketToSend>> PacketRouter::GeneratePadding(
+    size_t target_size_bytes) {
   rtc::CritScope cs(&modules_crit_);
   // First try on the last rtp module to have sent media. This increases the
   // the chance that any payload based padding will be useful as it will be
@@ -212,17 +219,17 @@ void PacketRouter::GeneratePadding(size_t target_size_bytes) {
     RTC_DCHECK(std::find(rtp_send_modules_.begin(), rtp_send_modules_.end(),
                          last_send_module_) != rtp_send_modules_.end());
     RTC_DCHECK(last_send_module_->HasBweExtensions());
-    last_send_module_->GeneratePadding(target_size_bytes);
+    return last_send_module_->GeneratePadding(target_size_bytes);
-    return;
   }
 
   // Rtp modules are ordered by which stream can most benefit from padding.
   for (RtpRtcp* rtp_module : rtp_send_modules_) {
     if (rtp_module->SendingMedia() && rtp_module->HasBweExtensions()) {
-      rtp_module->GeneratePadding(target_size_bytes);
+      return rtp_module->GeneratePadding(target_size_bytes);
-      return;
     }
   }
+
+  return {};
 }
 
 void PacketRouter::SetTransportWideSequenceNumber(uint16_t sequence_number) {

modules/pacing/packet_router.h

@@ -65,7 +65,8 @@ class PacketRouter : public TransportSequenceNumberAllocator,
   virtual size_t TimeToSendPadding(size_t bytes,
                                    const PacedPacketInfo& packet_info);
 
-  virtual void GeneratePadding(size_t target_size_bytes);
+  virtual std::vector<std::unique_ptr<RtpPacketToSend>> GeneratePadding(
+      size_t target_size_bytes);
 
   void SetTransportWideSequenceNumber(uint16_t sequence_number);
   uint16_t AllocateSequenceNumber() override;

modules/pacing/packet_router_unittest.cc

@@ -10,9 +10,12 @@
 
 #include <cstddef>
 #include <cstdint>
+#include <utility>
 
+#include "absl/memory/memory.h"
 #include "api/units/time_delta.h"
 #include "modules/pacing/packet_router.h"
+#include "modules/rtp_rtcp/include/rtp_header_extension_map.h"
 #include "modules/rtp_rtcp/mocks/mock_rtp_rtcp.h"
 #include "modules/rtp_rtcp/source/rtcp_packet/transport_feedback.h"
 #include "rtc_base/checks.h"
@@ -36,6 +39,7 @@ using ::testing::Field;
 using ::testing::Gt;
 using ::testing::Le;
 using ::testing::NiceMock;
+using ::testing::Property;
 using ::testing::Return;
 using ::testing::ReturnPointee;
 using ::testing::SaveArg;
@@ -296,9 +300,15 @@ TEST(PacketRouterTest, GeneratePaddingPicksCorrectModule) {
   packet_router.AddSendRtpModule(&rtp_2, false);
 
   const size_t kPaddingSize = 123;
+  const size_t kExpectedPaddingPackets = 1;
   EXPECT_CALL(rtp_1, GeneratePadding(_)).Times(0);
-  EXPECT_CALL(rtp_2, GeneratePadding(kPaddingSize)).Times(1);
+  EXPECT_CALL(rtp_2, GeneratePadding(kPaddingSize))
-  packet_router.GeneratePadding(kPaddingSize);
+      .WillOnce([&](size_t padding_size) {
+        return std::vector<std::unique_ptr<RtpPacketToSend>>(
+            kExpectedPaddingPackets);
+      });
+  auto generated_padding = packet_router.GeneratePadding(kPaddingSize);
+  EXPECT_EQ(generated_padding.size(), kExpectedPaddingPackets);
 
   packet_router.RemoveSendRtpModule(&rtp_1);
   packet_router.RemoveSendRtpModule(&rtp_2);
@@ -938,4 +948,86 @@ TEST(PacketRouterRembTest, ReceiveModuleTakesOverWhenLastSendModuleRemoved) {
   packet_router.RemoveReceiveRtpModule(&receive_module);
 }
 
+TEST(PacketRouterTest, SendPacketWithoutTransportSequenceNumbers) {
+  PacketRouter packet_router;
+  NiceMock<MockRtpRtcp> rtp_1;
+  packet_router.AddSendRtpModule(&rtp_1, false);
+
+  const uint16_t kSsrc1 = 1234;
+  ON_CALL(rtp_1, SendingMedia).WillByDefault(Return(true));
+  ON_CALL(rtp_1, SSRC).WillByDefault(Return(kSsrc1));
+
+  // Send a packet without TransportSequenceNumber extension registered,
+  // packets sent should not have the extension set.
+  RtpHeaderExtensionMap extension_manager;
+  auto packet = absl::make_unique<RtpPacketToSend>(&extension_manager);
+  packet->SetSsrc(kSsrc1);
+  EXPECT_CALL(
+      rtp_1,
+      TrySendPacket(
+          Property(&RtpPacketToSend::HasExtension<TransportSequenceNumber>,
+                   false),
+          _))
+      .WillOnce(Return(true));
+  packet_router.SendPacket(std::move(packet), PacedPacketInfo());
+
+  packet_router.RemoveSendRtpModule(&rtp_1);
+}
+
+TEST(PacketRouterTest, SendPacketAssignsTransportSequenceNumbers) {
+  PacketRouter packet_router;
+  NiceMock<MockRtpRtcp> rtp_1;
+  NiceMock<MockRtpRtcp> rtp_2;
+
+  packet_router.AddSendRtpModule(&rtp_1, false);
+  packet_router.AddSendRtpModule(&rtp_2, false);
+
+  const uint16_t kSsrc1 = 1234;
+  const uint16_t kSsrc2 = 2345;
+
+  ON_CALL(rtp_1, SendingMedia).WillByDefault(Return(true));
+  ON_CALL(rtp_1, SSRC).WillByDefault(Return(kSsrc1));
+  ON_CALL(rtp_2, SendingMedia).WillByDefault(Return(true));
+  ON_CALL(rtp_2, SSRC).WillByDefault(Return(kSsrc2));
+
+  RtpHeaderExtensionMap extension_manager;
+  const int kTransportSequenceNumberExtensionId = 1;
+  extension_manager.Register(kRtpExtensionTransportSequenceNumber,
+                             kTransportSequenceNumberExtensionId);
+
+  uint16_t transport_sequence_number = 0;
+
+  auto packet = absl::make_unique<RtpPacketToSend>(&extension_manager);
+  EXPECT_TRUE(packet->ReserveExtension<TransportSequenceNumber>());
+  packet->SetSsrc(kSsrc1);
+  EXPECT_CALL(
+      rtp_1,
+      TrySendPacket(
+          Property(&RtpPacketToSend::GetExtension<TransportSequenceNumber>,
+                   transport_sequence_number),
+          _))
+      .WillOnce(Return(true));
+  packet_router.SendPacket(std::move(packet), PacedPacketInfo());
+
+  ++transport_sequence_number;
+  packet = absl::make_unique<RtpPacketToSend>(&extension_manager);
+  EXPECT_TRUE(packet->ReserveExtension<TransportSequenceNumber>());
+  packet->SetSsrc(kSsrc2);
+
+  // There will be a failed attempt to send on kSsrc1 before trying
+  // the correct RTP module.
+  EXPECT_CALL(rtp_1, TrySendPacket).WillOnce(Return(false));
+  EXPECT_CALL(
+      rtp_2,
+      TrySendPacket(
+          Property(&RtpPacketToSend::GetExtension<TransportSequenceNumber>,
+                   transport_sequence_number),
+          _))
+      .WillOnce(Return(true));
+  packet_router.SendPacket(std::move(packet), PacedPacketInfo());
+
+  packet_router.RemoveSendRtpModule(&rtp_1);
+  packet_router.RemoveSendRtpModule(&rtp_2);
+}
+
 }  // namespace webrtc

modules/rtp_rtcp/include/rtp_rtcp.h

@@ -287,7 +287,8 @@ class RtpRtcp : public Module, public RtcpFeedbackSenderInterface {
   virtual size_t TimeToSendPadding(size_t bytes,
                                    const PacedPacketInfo& pacing_info) = 0;
 
-  virtual void GeneratePadding(size_t target_size_bytes) = 0;
+  virtual std::vector<std::unique_ptr<RtpPacketToSend>> GeneratePadding(
+      size_t target_size_bytes) = 0;
 
   // Called on generation of new statistics after an RTP send.
   virtual void RegisterSendChannelRtpStatisticsCallback(

modules/rtp_rtcp/mocks/mock_rtp_rtcp.h

@@ -95,7 +95,9 @@ class MockRtpRtcp : public RtpRtcp {
                     const PacedPacketInfo& pacing_info));
   MOCK_METHOD2(TimeToSendPadding,
                size_t(size_t bytes, const PacedPacketInfo& pacing_info));
-  MOCK_METHOD1(GeneratePadding, void(size_t target_size_bytes));
+  MOCK_METHOD1(
+      GeneratePadding,
+      std::vector<std::unique_ptr<RtpPacketToSend>>(size_t target_size_bytes));
   MOCK_METHOD2(RegisterRtcpObservers,
                void(RtcpIntraFrameObserver* intra_frame_callback,
                     RtcpBandwidthObserver* bandwidth_callback));

modules/rtp_rtcp/source/rtp_rtcp_impl.cc

@@ -409,8 +409,9 @@ size_t ModuleRtpRtcpImpl::TimeToSendPadding(
   return rtp_sender_->TimeToSendPadding(bytes, pacing_info);
 }
 
-void ModuleRtpRtcpImpl::GeneratePadding(size_t target_size_bytes) {
+std::vector<std::unique_ptr<RtpPacketToSend>>
-  rtp_sender_->GeneratePadding(target_size_bytes);
+ModuleRtpRtcpImpl::GeneratePadding(size_t target_size_bytes) {
+  return rtp_sender_->GeneratePadding(target_size_bytes);
 }
 
 size_t ModuleRtpRtcpImpl::MaxRtpPacketSize() const {

modules/rtp_rtcp/source/rtp_rtcp_impl.h

@@ -148,7 +148,8 @@ class ModuleRtpRtcpImpl : public RtpRtcp, public RTCPReceiver::ModuleRtpRtcp {
   size_t TimeToSendPadding(size_t bytes,
                            const PacedPacketInfo& pacing_info) override;
 
-  void GeneratePadding(size_t target_size_bytes) override;
+  std::vector<std::unique_ptr<RtpPacketToSend>> GeneratePadding(
+      size_t target_size_bytes) override;
 
   // RTCP part.
 

modules/rtp_rtcp/source/rtp_sender.cc

@@ -195,6 +195,9 @@ RTPSender::RTPSender(const RtpRtcp::Configuration& config)
                     config.field_trials)),
       payload_padding_prefer_useful_packets_(
           !IsDisabled("WebRTC-PayloadPadding-UseMostUsefulPacket",
+                      config.field_trials)),
+      pacer_legacy_packet_referencing_(
+          !IsDisabled("WebRTC-Pacer-LegacyPacketReferencing",
                       config.field_trials)) {
   // This random initialization is not intended to be cryptographic strong.
   timestamp_offset_ = random_.Rand<uint32_t>();
@@ -283,6 +286,9 @@ RTPSender::RTPSender(
               .find("Enabled") == 0),
       payload_padding_prefer_useful_packets_(
           field_trials.Lookup("WebRTC-PayloadPadding-UseMostUsefulPacket")
+              .find("Disabled") != 0),
+      pacer_legacy_packet_referencing_(
+          field_trials.Lookup("WebRTC-Pacer-LegacyPacketReferencing")
               .find("Disabled") != 0) {
   // This random initialization is not intended to be cryptographic strong.
   timestamp_offset_ = random_.Rand<uint32_t>();
@@ -592,31 +598,67 @@ int32_t RTPSender::ReSendPacket(uint16_t packet_id) {
   }
 
   const int32_t packet_size = static_cast<int32_t>(stored_packet->packet_size);
-
+  const bool rtx = (RtxStatus() & kRtxRetransmitted) > 0;
-  // Skip retransmission rate check if not configured.
-  if (retransmission_rate_limiter_) {
-    // Check if we're overusing retransmission bitrate.
-    // TODO(sprang): Add histograms for nack success or failure reasons.
-    if (!retransmission_rate_limiter_->TryUseRate(packet_size)) {
-      return -1;
-    }
-  }
 
   if (paced_sender_) {
-    // Mark packet as being in pacer queue again, to prevent duplicates.
+    if (pacer_legacy_packet_referencing_) {
-    if (!packet_history_.SetPendingTransmission(packet_id)) {
+      // Check if we're overusing retransmission bitrate.
-      // Packet has already been removed from history, return early.
+      // TODO(sprang): Add histograms for nack success or failure reasons.
-      return 0;
+      if (retransmission_rate_limiter_ &&
+          !retransmission_rate_limiter_->TryUseRate(packet_size)) {
+        return -1;
+      }
+
+      // Mark packet as being in pacer queue again, to prevent duplicates.
+      if (!packet_history_.SetPendingTransmission(packet_id)) {
+        // Packet has already been removed from history, return early.
+        return 0;
+      }
+
+      paced_sender_->InsertPacket(
+          RtpPacketSender::kNormalPriority, stored_packet->ssrc,
+          stored_packet->rtp_sequence_number, stored_packet->capture_time_ms,
+          stored_packet->packet_size, true);
+    } else {
+      std::unique_ptr<RtpPacketToSend> packet =
+          packet_history_.GetPacketAndMarkAsPending(
+              packet_id, [&](const RtpPacketToSend& stored_packet) {
+                // Check if we're overusing retransmission bitrate.
+                // TODO(sprang): Add histograms for nack success or failure
+                // reasons.
+                std::unique_ptr<RtpPacketToSend> retransmit_packet;
+                if (retransmission_rate_limiter_ &&
+                    !retransmission_rate_limiter_->TryUseRate(packet_size)) {
+                  return retransmit_packet;
+                }
+                if (rtx) {
+                  retransmit_packet = BuildRtxPacket(stored_packet);
+                } else {
+                  retransmit_packet =
+                      absl::make_unique<RtpPacketToSend>(stored_packet);
+                }
+                retransmit_packet->set_retransmitted_sequence_number(
+                    stored_packet.SequenceNumber());
+                return retransmit_packet;
+              });
+      if (!packet) {
+        return -1;
+      }
+      packet->set_packet_type(RtpPacketToSend::Type::kRetransmission);
+      paced_sender_->EnqueuePacket(std::move(packet));
     }
 
-    paced_sender_->InsertPacket(
-        RtpPacketSender::kNormalPriority, stored_packet->ssrc,
-        stored_packet->rtp_sequence_number, stored_packet->capture_time_ms,
-        stored_packet->packet_size, true);
-
     return packet_size;
   }
 
+  // TODO(sprang): Replace this whole code-path with a pass-through pacer.
+  // Check if we're overusing retransmission bitrate.
+  // TODO(sprang): Add histograms for nack success or failure reasons.
+  if (retransmission_rate_limiter_ &&
+      !retransmission_rate_limiter_->TryUseRate(packet_size)) {
+    return -1;
+  }
+
   std::unique_ptr<RtpPacketToSend> packet =
       packet_history_.GetPacketAndSetSendTime(packet_id);
   if (!packet) {
@@ -624,7 +666,6 @@ int32_t RTPSender::ReSendPacket(uint16_t packet_id) {
     return 0;
   }
 
-  const bool rtx = (RtxStatus() & kRtxRetransmitted) > 0;
   if (!PrepareAndSendPacket(std::move(packet), rtx, true, PacedPacketInfo()))
     return -1;
 
@@ -926,15 +967,18 @@ size_t RTPSender::TimeToSendPadding(size_t bytes,
   return bytes_sent;
 }
 
-void RTPSender::GeneratePadding(size_t target_size_bytes) {
+std::vector<std::unique_ptr<RtpPacketToSend>> RTPSender::GeneratePadding(
+    size_t target_size_bytes) {
   // This method does not actually send packets, it just generates
   // them and puts them in the pacer queue. Since this should incur
   // low overhead, keep the lock for the scope of the method in order
   // to make the code more readable.
   rtc::CritScope lock(&send_critsect_);
-  if (!sending_media_)
+  if (!sending_media_) {
-    return;
+    return {};
+  }
 
+  std::vector<std::unique_ptr<RtpPacketToSend>> padding_packets;
   size_t bytes_left = target_size_bytes;
   if ((rtx_ & kRtxRedundantPayloads) != 0) {
     while (bytes_left >= 0) {
@@ -953,7 +997,7 @@ void RTPSender::GeneratePadding(size_t target_size_bytes) {
 
       bytes_left -= std::min(bytes_left, packet->payload_size());
       packet->set_packet_type(RtpPacketToSend::Type::kPadding);
-      paced_sender_->EnqueuePacket(std::move(packet));
+      padding_packets.push_back(std::move(packet));
     }
   }
 
@@ -1022,10 +1066,15 @@ void RTPSender::GeneratePadding(size_t target_size_bytes) {
       padding_packet->SetPayloadType(rtx_payload_type_map_.begin()->second);
     }
 
+    if (rtp_header_extension_map_.IsRegistered(TransportSequenceNumber::kId)) {
+      padding_packet->ReserveExtension<TransportSequenceNumber>();
+    }
     padding_packet->SetPadding(padding_bytes_in_packet);
     bytes_left -= std::min(bytes_left, padding_bytes_in_packet);
-    paced_sender_->EnqueuePacket(std::move(padding_packet));
+    padding_packets.push_back(std::move(padding_packet));
   }
+
+  return padding_packets;
 }
 
 bool RTPSender::SendToNetwork(std::unique_ptr<RtpPacketToSend> packet,
@@ -1040,18 +1089,28 @@ bool RTPSender::SendToNetwork(std::unique_ptr<RtpPacketToSend> packet,
     size_t packet_size =
         send_side_bwe_with_overhead_ ? packet->size() : packet->payload_size();
     auto packet_type = packet->packet_type();
-    RTC_DCHECK(packet_type.has_value());
+    RTC_CHECK(packet_type) << "Packet type must be set before sending.";
-    if (ssrc == FlexfecSsrc()) {
+
-      // Store FlexFEC packets in the history here, so they can be found
+    if (pacer_legacy_packet_referencing_) {
-      // when the pacer calls TimeToSendPacket.
+      // If |pacer_reference_packets_| then pacer needs to find the packet in
-      flexfec_packet_history_.PutRtpPacket(std::move(packet), storage,
+      // the history when it is time to send, so move packet there.
-                                           absl::nullopt);
+      if (ssrc == FlexfecSsrc()) {
+        // Store FlexFEC packets in a separate history since they are on a
+        // separate SSRC.
+        flexfec_packet_history_.PutRtpPacket(std::move(packet), storage,
+                                             absl::nullopt);
+      } else {
+        packet_history_.PutRtpPacket(std::move(packet), storage, absl::nullopt);
+      }
+
+      paced_sender_->InsertPacket(PacketTypeToPriority(*packet_type), ssrc,
+                                  seq_no, capture_time_ms, packet_size, false);
     } else {
-      packet_history_.PutRtpPacket(std::move(packet), storage, absl::nullopt);
+      packet->set_allow_retransmission(storage ==
+                                       StorageType::kAllowRetransmission);
+      paced_sender_->EnqueuePacket(std::move(packet));
     }
 
-    paced_sender_->InsertPacket(PacketTypeToPriority(*packet_type), ssrc,
-                                seq_no, capture_time_ms, packet_size, false);
     return true;
   }
 

modules/rtp_rtcp/source/rtp_sender.h

@@ -119,7 +119,8 @@ class RTPSender {
   bool TrySendPacket(RtpPacketToSend* packet,
                      const PacedPacketInfo& pacing_info);
   size_t TimeToSendPadding(size_t bytes, const PacedPacketInfo& pacing_info);
-  void GeneratePadding(size_t target_size_bytes);
+  std::vector<std::unique_ptr<RtpPacketToSend>> GeneratePadding(
+      size_t target_size_bytes);
 
   // NACK.
   void OnReceivedNack(const std::vector<uint16_t>& nack_sequence_numbers,
@@ -322,6 +323,11 @@ class RTPSender {
   // packet_history_.GetBestFittingPacket() in TrySendRedundantPayloads().
   const bool payload_padding_prefer_useful_packets_;
 
+  // If true, PacedSender should only reference packets as in legacy mode.
+  // If false, PacedSender may have direct ownership of RtpPacketToSend objects.
+  // Defaults to true, will be changed to default false soon.
+  const bool pacer_legacy_packet_referencing_;
+
   RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(RTPSender);
 };