Adding upstream version 110.0.1.upstream/110.0.1upstream

Signed-off-by: Daniel Baumann <daniel.baumann@progress-linux.org>

1 files changed, 335 insertions, 0 deletions

diff --git a/third_party/libwebrtc/modules/pacing/packet_router.cc b/third_party/libwebrtc/modules/pacing/packet_router.cc
new file mode 100644
index 0000000000..a09f191bbd
--- /dev/null
+++ b/third_party/libwebrtc/modules/pacing/packet_router.cc
@@ -0,0 +1,335 @@
+/*
+ *  Copyright (c) 2015 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.
+ */
+
+#include "modules/pacing/packet_router.h"
+
+#include <algorithm>
+#include <cstdint>
+#include <limits>
+#include <memory>
+#include <utility>
+
+#include "absl/types/optional.h"
+#include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
+#include "modules/rtp_rtcp/source/rtcp_packet.h"
+#include "modules/rtp_rtcp/source/rtcp_packet/transport_feedback.h"
+#include "modules/rtp_rtcp/source/rtp_rtcp_interface.h"
+#include "rtc_base/checks.h"
+#include "rtc_base/logging.h"
+#include "rtc_base/system/unused.h"
+#include "rtc_base/time_utils.h"
+#include "rtc_base/trace_event.h"
+
+namespace webrtc {
+
+PacketRouter::PacketRouter() : PacketRouter(0) {}
+
+PacketRouter::PacketRouter(uint16_t start_transport_seq)
+    : last_send_module_(nullptr),
+      active_remb_module_(nullptr),
+      transport_seq_(start_transport_seq) {}
+
+PacketRouter::~PacketRouter() {
+  RTC_DCHECK(send_modules_map_.empty());
+  RTC_DCHECK(send_modules_list_.empty());
+  RTC_DCHECK(rtcp_feedback_senders_.empty());
+  RTC_DCHECK(sender_remb_candidates_.empty());
+  RTC_DCHECK(receiver_remb_candidates_.empty());
+  RTC_DCHECK(active_remb_module_ == nullptr);
+}
+
+void PacketRouter::AddSendRtpModule(RtpRtcpInterface* rtp_module,
+                                    bool remb_candidate) {
+  MutexLock lock(&modules_mutex_);
+
+  AddSendRtpModuleToMap(rtp_module, rtp_module->SSRC());
+  if (absl::optional<uint32_t> rtx_ssrc = rtp_module->RtxSsrc()) {
+    AddSendRtpModuleToMap(rtp_module, *rtx_ssrc);
+  }
+  if (absl::optional<uint32_t> flexfec_ssrc = rtp_module->FlexfecSsrc()) {
+    AddSendRtpModuleToMap(rtp_module, *flexfec_ssrc);
+  }
+
+  if (rtp_module->SupportsRtxPayloadPadding()) {
+    last_send_module_ = rtp_module;
+  }
+
+  if (remb_candidate) {
+    AddRembModuleCandidate(rtp_module, /* media_sender = */ true);
+  }
+}
+
+void PacketRouter::AddSendRtpModuleToMap(RtpRtcpInterface* rtp_module,
+                                         uint32_t ssrc) {
+  RTC_DCHECK(send_modules_map_.find(ssrc) == send_modules_map_.end());
+
+  // Signal to module that the pacer thread is attached and can send packets.
+  rtp_module->OnPacketSendingThreadSwitched();
+
+  // Always keep the audio modules at the back of the list, so that when we
+  // iterate over the modules in order to find one that can send padding we
+  // will prioritize video. This is important to make sure they are counted
+  // into the bandwidth estimate properly.
+  if (rtp_module->IsAudioConfigured()) {
+    send_modules_list_.push_back(rtp_module);
+  } else {
+    send_modules_list_.push_front(rtp_module);
+  }
+  send_modules_map_[ssrc] = rtp_module;
+}
+
+void PacketRouter::RemoveSendRtpModuleFromMap(uint32_t ssrc) {
+  auto kv = send_modules_map_.find(ssrc);
+  RTC_DCHECK(kv != send_modules_map_.end());
+  send_modules_list_.remove(kv->second);
+  send_modules_map_.erase(kv);
+}
+
+void PacketRouter::RemoveSendRtpModule(RtpRtcpInterface* rtp_module) {
+  MutexLock lock(&modules_mutex_);
+  MaybeRemoveRembModuleCandidate(rtp_module, /* media_sender = */ true);
+
+  RemoveSendRtpModuleFromMap(rtp_module->SSRC());
+  if (absl::optional<uint32_t> rtx_ssrc = rtp_module->RtxSsrc()) {
+    RemoveSendRtpModuleFromMap(*rtx_ssrc);
+  }
+  if (absl::optional<uint32_t> flexfec_ssrc = rtp_module->FlexfecSsrc()) {
+    RemoveSendRtpModuleFromMap(*flexfec_ssrc);
+  }
+
+  if (last_send_module_ == rtp_module) {
+    last_send_module_ = nullptr;
+  }
+  rtp_module->OnPacketSendingThreadSwitched();
+}
+
+void PacketRouter::AddReceiveRtpModule(RtcpFeedbackSenderInterface* rtcp_sender,
+                                       bool remb_candidate) {
+  MutexLock lock(&modules_mutex_);
+  RTC_DCHECK(std::find(rtcp_feedback_senders_.begin(),
+                       rtcp_feedback_senders_.end(),
+                       rtcp_sender) == rtcp_feedback_senders_.end());
+
+  rtcp_feedback_senders_.push_back(rtcp_sender);
+
+  if (remb_candidate) {
+    AddRembModuleCandidate(rtcp_sender, /* media_sender = */ false);
+  }
+}
+
+void PacketRouter::RemoveReceiveRtpModule(
+    RtcpFeedbackSenderInterface* rtcp_sender) {
+  MutexLock lock(&modules_mutex_);
+  MaybeRemoveRembModuleCandidate(rtcp_sender, /* media_sender = */ false);
+  auto it = std::find(rtcp_feedback_senders_.begin(),
+                      rtcp_feedback_senders_.end(), rtcp_sender);
+  RTC_DCHECK(it != rtcp_feedback_senders_.end());
+  rtcp_feedback_senders_.erase(it);
+}
+
+void PacketRouter::SendPacket(std::unique_ptr<RtpPacketToSend> packet,
+                              const PacedPacketInfo& cluster_info) {
+  TRACE_EVENT2(TRACE_DISABLED_BY_DEFAULT("webrtc"), "PacketRouter::SendPacket",
+               "sequence_number", packet->SequenceNumber(), "rtp_timestamp",
+               packet->Timestamp());
+
+  MutexLock lock(&modules_mutex_);
+  // With the new pacer code path, transport sequence numbers are only set here,
+  // on the pacer thread. Therefore we don't need atomics/synchronization.
+  bool assign_transport_sequence_number =
+      packet->HasExtension<TransportSequenceNumber>();
+  if (assign_transport_sequence_number) {
+    packet->SetExtension<TransportSequenceNumber>((transport_seq_ + 1) &
+                                                  0xFFFF);
+  }
+
+  uint32_t ssrc = packet->Ssrc();
+  auto kv = send_modules_map_.find(ssrc);
+  if (kv == send_modules_map_.end()) {
+    RTC_LOG(LS_WARNING)
+        << "Failed to send packet, matching RTP module not found "
+           "or transport error. SSRC = "
+        << packet->Ssrc() << ", sequence number " << packet->SequenceNumber();
+    return;
+  }
+
+  RtpRtcpInterface* rtp_module = kv->second;
+  if (!rtp_module->TrySendPacket(packet.get(), cluster_info)) {
+    RTC_LOG(LS_WARNING) << "Failed to send packet, rejected by RTP module.";
+    return;
+  }
+
+  // Sending succeeded.
+
+  if (assign_transport_sequence_number) {
+    ++transport_seq_;
+  }
+
+  if (rtp_module->SupportsRtxPayloadPadding()) {
+    // This is now the last module to send media, and has the desired
+    // properties needed for payload based padding. Cache it for later use.
+    last_send_module_ = rtp_module;
+  }
+
+  for (auto& packet : rtp_module->FetchFecPackets()) {
+    pending_fec_packets_.push_back(std::move(packet));
+  }
+}
+
+std::vector<std::unique_ptr<RtpPacketToSend>> PacketRouter::FetchFec() {
+  MutexLock lock(&modules_mutex_);
+  std::vector<std::unique_ptr<RtpPacketToSend>> fec_packets =
+      std::move(pending_fec_packets_);
+  pending_fec_packets_.clear();
+  return fec_packets;
+}
+
+std::vector<std::unique_ptr<RtpPacketToSend>> PacketRouter::GeneratePadding(
+    DataSize size) {
+  TRACE_EVENT1(TRACE_DISABLED_BY_DEFAULT("webrtc"),
+               "PacketRouter::GeneratePadding", "bytes", size.bytes());
+
+  MutexLock lock(&modules_mutex_);
+  // 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
+  // somewhat distributed over modules according the packet rate, even if it
+  // will be more skewed towards the highest bitrate stream. At the very least
+  // this prevents sending payload padding on a disabled stream where it's
+  // guaranteed not to be useful.
+  std::vector<std::unique_ptr<RtpPacketToSend>> padding_packets;
+  if (last_send_module_ != nullptr &&
+      last_send_module_->SupportsRtxPayloadPadding()) {
+    padding_packets = last_send_module_->GeneratePadding(size.bytes());
+  }
+
+  if (padding_packets.empty()) {
+    // Iterate over all modules send module. Video modules will be at the front
+    // and so will be prioritized. This is important since audio packets may not
+    // be taken into account by the bandwidth estimator, e.g. in FF.
+    for (RtpRtcpInterface* rtp_module : send_modules_list_) {
+      if (rtp_module->SupportsPadding()) {
+        padding_packets = rtp_module->GeneratePadding(size.bytes());
+        if (!padding_packets.empty()) {
+          last_send_module_ = rtp_module;
+          break;
+        }
+      }
+    }
+  }
+
+  for (auto& packet : padding_packets) {
+    RTC_UNUSED(packet);
+    TRACE_EVENT2(TRACE_DISABLED_BY_DEFAULT("webrtc"),
+                 "PacketRouter::GeneratePadding::Loop", "sequence_number",
+                 packet->SequenceNumber(), "rtp_timestamp",
+                 packet->Timestamp());
+  }
+
+  return padding_packets;
+}
+
+uint16_t PacketRouter::CurrentTransportSequenceNumber() const {
+  MutexLock lock(&modules_mutex_);
+  return transport_seq_ & 0xFFFF;
+}
+
+void PacketRouter::SendRemb(int64_t bitrate_bps, std::vector<uint32_t> ssrcs) {
+  MutexLock lock(&modules_mutex_);
+
+  if (!active_remb_module_) {
+    return;
+  }
+
+  // The Add* and Remove* methods above ensure that REMB is disabled on all
+  // other modules, because otherwise, they will send REMB with stale info.
+  active_remb_module_->SetRemb(bitrate_bps, std::move(ssrcs));
+}
+
+void PacketRouter::SendCombinedRtcpPacket(
+    std::vector<std::unique_ptr<rtcp::RtcpPacket>> packets) {
+  MutexLock lock(&modules_mutex_);
+
+  // Prefer send modules.
+  for (RtpRtcpInterface* rtp_module : send_modules_list_) {
+    if (rtp_module->RTCP() == RtcpMode::kOff) {
+      continue;
+    }
+    rtp_module->SendCombinedRtcpPacket(std::move(packets));
+    return;
+  }
+
+  if (rtcp_feedback_senders_.empty()) {
+    return;
+  }
+  auto* rtcp_sender = rtcp_feedback_senders_[0];
+  rtcp_sender->SendCombinedRtcpPacket(std::move(packets));
+}
+
+void PacketRouter::AddRembModuleCandidate(
+    RtcpFeedbackSenderInterface* candidate_module,
+    bool media_sender) {
+  RTC_DCHECK(candidate_module);
+  std::vector<RtcpFeedbackSenderInterface*>& candidates =
+      media_sender ? sender_remb_candidates_ : receiver_remb_candidates_;
+  RTC_DCHECK(std::find(candidates.cbegin(), candidates.cend(),
+                       candidate_module) == candidates.cend());
+  candidates.push_back(candidate_module);
+  DetermineActiveRembModule();
+}
+
+void PacketRouter::MaybeRemoveRembModuleCandidate(
+    RtcpFeedbackSenderInterface* candidate_module,
+    bool media_sender) {
+  RTC_DCHECK(candidate_module);
+  std::vector<RtcpFeedbackSenderInterface*>& candidates =
+      media_sender ? sender_remb_candidates_ : receiver_remb_candidates_;
+  auto it = std::find(candidates.begin(), candidates.end(), candidate_module);
+
+  if (it == candidates.end()) {
+    return;  // Function called due to removal of non-REMB-candidate module.
+  }
+
+  if (*it == active_remb_module_) {
+    UnsetActiveRembModule();
+  }
+  candidates.erase(it);
+  DetermineActiveRembModule();
+}
+
+void PacketRouter::UnsetActiveRembModule() {
+  RTC_CHECK(active_remb_module_);
+  active_remb_module_->UnsetRemb();
+  active_remb_module_ = nullptr;
+}
+
+void PacketRouter::DetermineActiveRembModule() {
+  // Sender modules take precedence over receiver modules, because SRs (sender
+  // reports) are sent more frequently than RR (receiver reports).
+  // When adding the first sender module, we should change the active REMB
+  // module to be that. Otherwise, we remain with the current active module.
+
+  RtcpFeedbackSenderInterface* new_active_remb_module;
+
+  if (!sender_remb_candidates_.empty()) {
+    new_active_remb_module = sender_remb_candidates_.front();
+  } else if (!receiver_remb_candidates_.empty()) {
+    new_active_remb_module = receiver_remb_candidates_.front();
+  } else {
+    new_active_remb_module = nullptr;
+  }
+
+  if (new_active_remb_module != active_remb_module_ && active_remb_module_) {
+    UnsetActiveRembModule();
+  }
+
+  active_remb_module_ = new_active_remb_module;
+}
+
+}  // namespace webrtc