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/*
* 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 <memory>
#include <optional>
#include <utility>
#include <vector>
#include "absl/functional/any_invocable.h"
#include "api/array_view.h"
#include "api/rtp_headers.h"
#include "api/sequence_checker.h"
#include "api/transport/network_types.h"
#include "api/units/data_size.h"
#include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
#include "modules/rtp_rtcp/source/rtcp_packet.h"
#include "modules/rtp_rtcp/source/rtp_header_extensions.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/trace_event.h"
namespace webrtc {
PacketRouter::PacketRouter()
: last_send_module_(nullptr),
active_remb_module_(nullptr),
transport_seq_(1) {}
PacketRouter::~PacketRouter() {
RTC_DCHECK_RUN_ON(&thread_checker_);
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) {
RTC_DCHECK_RUN_ON(&thread_checker_);
AddSendRtpModuleToMap(rtp_module, rtp_module->SSRC());
if (std::optional<uint32_t> rtx_ssrc = rtp_module->RtxSsrc()) {
AddSendRtpModuleToMap(rtp_module, *rtx_ssrc);
}
if (std::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);
}
}
bool PacketRouter::SupportsRtxPayloadPadding() const {
RTC_DCHECK_RUN_ON(&thread_checker_);
for (RtpRtcpInterface* rtp_module : send_modules_list_) {
if (rtp_module->SupportsRtxPayloadPadding()) {
return true;
}
}
return false;
}
void PacketRouter::RegisterNotifyBweCallback(
absl::AnyInvocable<void(const RtpPacketToSend& packet,
const PacedPacketInfo& pacing_info)> callback) {
RTC_DCHECK_RUN_ON(&thread_checker_);
notify_bwe_callback_ = std::move(callback);
}
void PacketRouter::AddSendRtpModuleToMap(RtpRtcpInterface* rtp_module,
uint32_t ssrc) {
RTC_DCHECK_RUN_ON(&thread_checker_);
RTC_CHECK(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) {
RTC_DCHECK_RUN_ON(&thread_checker_);
auto it = send_modules_map_.find(ssrc);
if (it == send_modules_map_.end()) {
RTC_LOG(LS_ERROR) << "No send module found for ssrc " << ssrc;
return;
}
send_modules_list_.remove(it->second);
RTC_CHECK(modules_used_in_current_batch_.empty());
send_modules_map_.erase(it);
}
void PacketRouter::RemoveSendRtpModule(RtpRtcpInterface* rtp_module) {
RTC_DCHECK_RUN_ON(&thread_checker_);
MaybeRemoveRembModuleCandidate(rtp_module, /* media_sender = */ true);
RemoveSendRtpModuleFromMap(rtp_module->SSRC());
if (std::optional<uint32_t> rtx_ssrc = rtp_module->RtxSsrc()) {
RemoveSendRtpModuleFromMap(*rtx_ssrc);
}
if (std::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) {
RTC_DCHECK_RUN_ON(&thread_checker_);
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) {
RTC_DCHECK_RUN_ON(&thread_checker_);
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) {
RTC_DCHECK_RUN_ON(&thread_checker_);
TRACE_EVENT2(TRACE_DISABLED_BY_DEFAULT("webrtc"), "PacketRouter::SendPacket",
"sequence_number", packet->SequenceNumber(), "rtp_timestamp",
packet->Timestamp());
uint32_t ssrc = packet->Ssrc();
auto it = send_modules_map_.find(ssrc);
if (it == 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 = it->second;
if (!packet || !rtp_module->CanSendPacket(*packet)) {
RTC_LOG(LS_WARNING) << "Failed to send packet, Not sending media";
return;
}
// TODO(bugs.webrtc.org/15368): Even if the TransportSequenceNumber extension
// is not negotiated, we will need the transport sequence number for BWE.
if (packet->HasExtension<TransportSequenceNumber>()) {
packet->set_transport_sequence_number(transport_seq_++);
}
rtp_module->AssignSequenceNumber(*packet);
if (notify_bwe_callback_) {
notify_bwe_callback_(*packet, cluster_info);
}
rtp_module->SendPacket(std::move(packet), cluster_info);
modules_used_in_current_batch_.insert(rtp_module);
// Sending succeeded.
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));
}
}
void PacketRouter::OnBatchComplete() {
RTC_DCHECK_RUN_ON(&thread_checker_);
TRACE_EVENT0(TRACE_DISABLED_BY_DEFAULT("webrtc"),
"PacketRouter::OnBatchComplete");
for (auto& module : modules_used_in_current_batch_) {
module->OnBatchComplete();
}
modules_used_in_current_batch_.clear();
}
std::vector<std::unique_ptr<RtpPacketToSend>> PacketRouter::FetchFec() {
RTC_DCHECK_RUN_ON(&thread_checker_);
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) {
RTC_DCHECK_RUN_ON(&thread_checker_);
TRACE_EVENT1(TRACE_DISABLED_BY_DEFAULT("webrtc"),
"PacketRouter::GeneratePadding", "bytes", size.bytes());
// 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;
}
void PacketRouter::OnAbortedRetransmissions(
uint32_t ssrc,
rtc::ArrayView<const uint16_t> sequence_numbers) {
RTC_DCHECK_RUN_ON(&thread_checker_);
auto it = send_modules_map_.find(ssrc);
if (it != send_modules_map_.end()) {
it->second->OnAbortedRetransmissions(sequence_numbers);
}
}
std::optional<uint32_t> PacketRouter::GetRtxSsrcForMedia(uint32_t ssrc) const {
RTC_DCHECK_RUN_ON(&thread_checker_);
auto it = send_modules_map_.find(ssrc);
if (it != send_modules_map_.end() && it->second->SSRC() == ssrc) {
// A module is registered with the given SSRC, and that SSRC is the main
// media SSRC for that RTP module.
return it->second->RtxSsrc();
}
return std::nullopt;
}
void PacketRouter::SendRemb(int64_t bitrate_bps, std::vector<uint32_t> ssrcs) {
RTC_DCHECK_RUN_ON(&thread_checker_);
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) {
RTC_DCHECK_RUN_ON(&thread_checker_);
// 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_RUN_ON(&thread_checker_);
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_DCHECK_RUN_ON(&thread_checker_);
RTC_CHECK(active_remb_module_);
active_remb_module_->UnsetRemb();
active_remb_module_ = nullptr;
}
void PacketRouter::DetermineActiveRembModule() {
RTC_DCHECK_RUN_ON(&thread_checker_);
// 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