| /* |
| * 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 <limits> |
| |
| #include "modules/rtp_rtcp/include/rtp_rtcp.h" |
| #include "modules/rtp_rtcp/include/rtp_rtcp_defines.h" |
| #include "modules/rtp_rtcp/source/rtcp_packet/transport_feedback.h" |
| #include "rtc_base/atomicops.h" |
| #include "rtc_base/checks.h" |
| #include "rtc_base/timeutils.h" |
| |
| namespace webrtc { |
| namespace { |
| |
| constexpr int kRembSendIntervalMs = 200; |
| |
| } // namespace |
| |
| PacketRouter::PacketRouter() |
| : last_send_module_(nullptr), |
| last_remb_time_ms_(rtc::TimeMillis()), |
| last_send_bitrate_bps_(0), |
| bitrate_bps_(0), |
| max_bitrate_bps_(std::numeric_limits<decltype(max_bitrate_bps_)>::max()), |
| active_remb_module_(nullptr), |
| transport_seq_(0) {} |
| |
| PacketRouter::~PacketRouter() { |
| RTC_DCHECK(rtp_send_modules_.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(RtpRtcp* rtp_module, bool remb_candidate) { |
| rtc::CritScope cs(&modules_crit_); |
| RTC_DCHECK(std::find(rtp_send_modules_.begin(), rtp_send_modules_.end(), |
| rtp_module) == rtp_send_modules_.end()); |
| // Put modules which can use regular payload packets (over rtx) instead of |
| // padding first as it's less of a waste |
| if ((rtp_module->RtxSendStatus() & kRtxRedundantPayloads) > 0) { |
| rtp_send_modules_.push_front(rtp_module); |
| } else { |
| rtp_send_modules_.push_back(rtp_module); |
| } |
| |
| if (remb_candidate) { |
| AddRembModuleCandidate(rtp_module, /* media_sender = */ true); |
| } |
| } |
| |
| void PacketRouter::RemoveSendRtpModule(RtpRtcp* rtp_module) { |
| rtc::CritScope cs(&modules_crit_); |
| MaybeRemoveRembModuleCandidate(rtp_module, /* media_sender = */ true); |
| auto it = |
| std::find(rtp_send_modules_.begin(), rtp_send_modules_.end(), rtp_module); |
| RTC_DCHECK(it != rtp_send_modules_.end()); |
| rtp_send_modules_.erase(it); |
| if (last_send_module_ == rtp_module) { |
| last_send_module_ = nullptr; |
| } |
| } |
| |
| void PacketRouter::AddReceiveRtpModule(RtcpFeedbackSenderInterface* rtcp_sender, |
| bool remb_candidate) { |
| rtc::CritScope cs(&modules_crit_); |
| 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::CritScope cs(&modules_crit_); |
| 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); |
| } |
| |
| bool PacketRouter::TimeToSendPacket(uint32_t ssrc, |
| uint16_t sequence_number, |
| int64_t capture_timestamp, |
| bool retransmission, |
| const PacedPacketInfo& pacing_info) { |
| rtc::CritScope cs(&modules_crit_); |
| for (auto* rtp_module : rtp_send_modules_) { |
| if (!rtp_module->SendingMedia()) { |
| continue; |
| } |
| if (ssrc == rtp_module->SSRC() || ssrc == rtp_module->FlexfecSsrc()) { |
| if ((rtp_module->RtxSendStatus() & kRtxRedundantPayloads) && |
| rtp_module->HasBweExtensions()) { |
| // 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; |
| } |
| return rtp_module->TimeToSendPacket(ssrc, sequence_number, |
| capture_timestamp, retransmission, |
| pacing_info); |
| } |
| } |
| return true; |
| } |
| |
| size_t PacketRouter::TimeToSendPadding(size_t bytes_to_send, |
| const PacedPacketInfo& pacing_info) { |
| size_t total_bytes_sent = 0; |
| 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 |
| // 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. |
| if (last_send_module_ != nullptr) { |
| 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()); |
| total_bytes_sent += last_send_module_->TimeToSendPadding( |
| bytes_to_send - total_bytes_sent, pacing_info); |
| if (total_bytes_sent >= bytes_to_send) { |
| return total_bytes_sent; |
| } |
| } |
| |
| // Rtp modules are ordered by which stream can most benefit from padding. |
| for (RtpRtcp* module : rtp_send_modules_) { |
| if (module->SendingMedia() && module->HasBweExtensions()) { |
| size_t bytes_sent = module->TimeToSendPadding( |
| bytes_to_send - total_bytes_sent, pacing_info); |
| total_bytes_sent += bytes_sent; |
| if (total_bytes_sent >= bytes_to_send) |
| break; |
| } |
| } |
| return total_bytes_sent; |
| } |
| |
| void PacketRouter::SetTransportWideSequenceNumber(uint16_t sequence_number) { |
| rtc::AtomicOps::ReleaseStore(&transport_seq_, sequence_number); |
| } |
| |
| uint16_t PacketRouter::AllocateSequenceNumber() { |
| int prev_seq = rtc::AtomicOps::AcquireLoad(&transport_seq_); |
| int desired_prev_seq; |
| int new_seq; |
| do { |
| desired_prev_seq = prev_seq; |
| new_seq = (desired_prev_seq + 1) & 0xFFFF; |
| // Note: CompareAndSwap returns the actual value of transport_seq at the |
| // time the CAS operation was executed. Thus, if prev_seq is returned, the |
| // operation was successful - otherwise we need to retry. Saving the |
| // return value saves us a load on retry. |
| prev_seq = rtc::AtomicOps::CompareAndSwap(&transport_seq_, desired_prev_seq, |
| new_seq); |
| } while (prev_seq != desired_prev_seq); |
| |
| return new_seq; |
| } |
| |
| void PacketRouter::OnReceiveBitrateChanged(const std::vector<uint32_t>& ssrcs, |
| uint32_t bitrate_bps) { |
| // % threshold for if we should send a new REMB asap. |
| const int64_t kSendThresholdPercent = 97; |
| // TODO(danilchap): Remove receive_bitrate_bps variable and the cast |
| // when OnReceiveBitrateChanged takes bitrate as int64_t. |
| int64_t receive_bitrate_bps = static_cast<int64_t>(bitrate_bps); |
| |
| int64_t now_ms = rtc::TimeMillis(); |
| { |
| rtc::CritScope lock(&remb_crit_); |
| |
| // If we already have an estimate, check if the new total estimate is below |
| // kSendThresholdPercent of the previous estimate. |
| if (last_send_bitrate_bps_ > 0) { |
| int64_t new_remb_bitrate_bps = |
| last_send_bitrate_bps_ - bitrate_bps_ + receive_bitrate_bps; |
| |
| if (new_remb_bitrate_bps < |
| kSendThresholdPercent * last_send_bitrate_bps_ / 100) { |
| // The new bitrate estimate is less than kSendThresholdPercent % of the |
| // last report. Send a REMB asap. |
| last_remb_time_ms_ = now_ms - kRembSendIntervalMs; |
| } |
| } |
| bitrate_bps_ = receive_bitrate_bps; |
| |
| if (now_ms - last_remb_time_ms_ < kRembSendIntervalMs) { |
| return; |
| } |
| // NOTE: Updated if we intend to send the data; we might not have |
| // a module to actually send it. |
| last_remb_time_ms_ = now_ms; |
| last_send_bitrate_bps_ = receive_bitrate_bps; |
| // Cap the value to send in remb with configured value. |
| receive_bitrate_bps = std::min(receive_bitrate_bps, max_bitrate_bps_); |
| } |
| SendRemb(receive_bitrate_bps, ssrcs); |
| } |
| |
| void PacketRouter::SetMaxDesiredReceiveBitrate(int64_t bitrate_bps) { |
| RTC_DCHECK_GE(bitrate_bps, 0); |
| { |
| rtc::CritScope lock(&remb_crit_); |
| max_bitrate_bps_ = bitrate_bps; |
| if (rtc::TimeMillis() - last_remb_time_ms_ < kRembSendIntervalMs && |
| last_send_bitrate_bps_ > 0 && |
| last_send_bitrate_bps_ <= max_bitrate_bps_) { |
| // Recent measured bitrate is already below the cap. |
| return; |
| } |
| } |
| SendRemb(bitrate_bps, /*ssrcs=*/{}); |
| } |
| |
| bool PacketRouter::SendRemb(int64_t bitrate_bps, |
| const std::vector<uint32_t>& ssrcs) { |
| rtc::CritScope lock(&modules_crit_); |
| |
| if (!active_remb_module_) { |
| return false; |
| } |
| |
| // 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, ssrcs); |
| |
| return true; |
| } |
| |
| bool PacketRouter::SendTransportFeedback(rtcp::TransportFeedback* packet) { |
| rtc::CritScope cs(&modules_crit_); |
| // Prefer send modules. |
| for (auto* rtp_module : rtp_send_modules_) { |
| packet->SetSenderSsrc(rtp_module->SSRC()); |
| if (rtp_module->SendFeedbackPacket(*packet)) { |
| return true; |
| } |
| } |
| for (auto* rtcp_sender : rtcp_feedback_senders_) { |
| packet->SetSenderSsrc(rtcp_sender->SSRC()); |
| if (rtcp_sender->SendFeedbackPacket(*packet)) { |
| return true; |
| } |
| } |
| return false; |
| } |
| |
| 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 |