From 26a029d407be480d791972afb5975cf62c9360a6 Mon Sep 17 00:00:00 2001 From: Daniel Baumann Date: Fri, 19 Apr 2024 02:47:55 +0200 Subject: Adding upstream version 124.0.1. Signed-off-by: Daniel Baumann --- .../modules/rtp_rtcp/source/rtp_rtcp_impl.cc | 730 +++++++++++++++++++++ 1 file changed, 730 insertions(+) create mode 100644 third_party/libwebrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.cc (limited to 'third_party/libwebrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.cc') diff --git a/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.cc b/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.cc new file mode 100644 index 0000000000..a63067141d --- /dev/null +++ b/third_party/libwebrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.cc @@ -0,0 +1,730 @@ +/* + * Copyright (c) 2012 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/rtp_rtcp/source/rtp_rtcp_impl.h" + +#include + +#include +#include +#include +#include +#include +#include + +#include "absl/strings/string_view.h" +#include "absl/types/optional.h" +#include "modules/rtp_rtcp/source/rtcp_packet/dlrr.h" +#include "modules/rtp_rtcp/source/rtcp_sender.h" +#include "modules/rtp_rtcp/source/rtp_rtcp_config.h" +#include "modules/rtp_rtcp/source/rtp_rtcp_interface.h" +#include "modules/rtp_rtcp/source/time_util.h" +#include "rtc_base/checks.h" +#include "rtc_base/logging.h" +#include "system_wrappers/include/ntp_time.h" + +#ifdef _WIN32 +// Disable warning C4355: 'this' : used in base member initializer list. +#pragma warning(disable : 4355) +#endif + +namespace webrtc { +namespace { +const int64_t kRtpRtcpRttProcessTimeMs = 1000; +const int64_t kRtpRtcpBitrateProcessTimeMs = 10; +constexpr TimeDelta kDefaultExpectedRetransmissionTime = TimeDelta::Millis(125); +} // namespace + +ModuleRtpRtcpImpl::RtpSenderContext::RtpSenderContext( + const RtpRtcpInterface::Configuration& config) + : packet_history(config.clock, RtpPacketHistory::PaddingMode::kPriority), + sequencer_(config.local_media_ssrc, + config.rtx_send_ssrc, + /*require_marker_before_media_padding=*/!config.audio, + config.clock), + packet_sender(config, &packet_history), + non_paced_sender(&packet_sender, &sequencer_), + packet_generator( + config, + &packet_history, + config.paced_sender ? config.paced_sender : &non_paced_sender) {} + +std::unique_ptr RtpRtcp::DEPRECATED_Create( + const Configuration& configuration) { + RTC_DCHECK(configuration.clock); + return std::make_unique(configuration); +} + +ModuleRtpRtcpImpl::ModuleRtpRtcpImpl(const Configuration& configuration) + : rtcp_sender_( + RTCPSender::Configuration::FromRtpRtcpConfiguration(configuration)), + rtcp_receiver_(configuration, this), + clock_(configuration.clock), + last_bitrate_process_time_(clock_->TimeInMilliseconds()), + last_rtt_process_time_(clock_->TimeInMilliseconds()), + packet_overhead_(28), // IPV4 UDP. + nack_last_time_sent_full_ms_(0), + nack_last_seq_number_sent_(0), + rtt_stats_(configuration.rtt_stats), + rtt_ms_(0) { + if (!configuration.receiver_only) { + rtp_sender_ = std::make_unique(configuration); + // Make sure rtcp sender use same timestamp offset as rtp sender. + rtcp_sender_.SetTimestampOffset( + rtp_sender_->packet_generator.TimestampOffset()); + } + + // Set default packet size limit. + // TODO(nisse): Kind-of duplicates + // webrtc::VideoSendStream::Config::Rtp::kDefaultMaxPacketSize. + const size_t kTcpOverIpv4HeaderSize = 40; + SetMaxRtpPacketSize(IP_PACKET_SIZE - kTcpOverIpv4HeaderSize); +} + +ModuleRtpRtcpImpl::~ModuleRtpRtcpImpl() = default; + +// Process any pending tasks such as timeouts (non time critical events). +void ModuleRtpRtcpImpl::Process() { + const int64_t now = clock_->TimeInMilliseconds(); + + if (rtp_sender_) { + if (now >= last_bitrate_process_time_ + kRtpRtcpBitrateProcessTimeMs) { + rtp_sender_->packet_sender.ProcessBitrateAndNotifyObservers(); + last_bitrate_process_time_ = now; + } + } + + // TODO(bugs.webrtc.org/11581): We update the RTT once a second, whereas other + // things that run in this method are updated much more frequently. Move the + // RTT checking over to the worker thread, which matches better with where the + // stats are maintained. + bool process_rtt = now >= last_rtt_process_time_ + kRtpRtcpRttProcessTimeMs; + if (rtcp_sender_.Sending()) { + // Process RTT if we have received a report block and we haven't + // processed RTT for at least `kRtpRtcpRttProcessTimeMs` milliseconds. + // Note that LastReceivedReportBlockMs() grabs a lock, so check + // `process_rtt` first. + if (process_rtt && rtt_stats_ != nullptr && + rtcp_receiver_.LastReceivedReportBlockMs() > last_rtt_process_time_) { + TimeDelta max_rtt = TimeDelta::Zero(); + for (const auto& block : rtcp_receiver_.GetLatestReportBlockData()) { + if (block.last_rtt() > max_rtt) { + max_rtt = block.last_rtt(); + } + } + // Report the rtt. + if (max_rtt > TimeDelta::Zero()) { + rtt_stats_->OnRttUpdate(max_rtt.ms()); + } + } + + // Verify receiver reports are delivered and the reported sequence number + // is increasing. + // TODO(bugs.webrtc.org/11581): The timeout value needs to be checked every + // few seconds (see internals of RtcpRrTimeout). Here, we may be polling it + // a couple of hundred times a second, which isn't great since it grabs a + // lock. Note also that LastReceivedReportBlockMs() (called above) and + // RtcpRrTimeout() both grab the same lock and check the same timer, so + // it should be possible to consolidate that work somehow. + if (rtcp_receiver_.RtcpRrTimeout()) { + RTC_LOG_F(LS_WARNING) << "Timeout: No RTCP RR received."; + } else if (rtcp_receiver_.RtcpRrSequenceNumberTimeout()) { + RTC_LOG_F(LS_WARNING) << "Timeout: No increase in RTCP RR extended " + "highest sequence number."; + } + } else { + // Report rtt from receiver. + if (process_rtt && rtt_stats_ != nullptr) { + absl::optional rtt = rtcp_receiver_.GetAndResetXrRrRtt(); + if (rtt.has_value()) { + rtt_stats_->OnRttUpdate(rtt->ms()); + } + } + } + + // Get processed rtt. + if (process_rtt) { + last_rtt_process_time_ = now; + if (rtt_stats_) { + // Make sure we have a valid RTT before setting. + int64_t last_rtt = rtt_stats_->LastProcessedRtt(); + if (last_rtt >= 0) + set_rtt_ms(last_rtt); + } + } + + if (rtcp_sender_.TimeToSendRTCPReport()) + rtcp_sender_.SendRTCP(GetFeedbackState(), kRtcpReport); + + if (rtcp_sender_.TMMBR() && rtcp_receiver_.UpdateTmmbrTimers()) { + rtcp_receiver_.NotifyTmmbrUpdated(); + } +} + +void ModuleRtpRtcpImpl::SetRtxSendStatus(int mode) { + rtp_sender_->packet_generator.SetRtxStatus(mode); +} + +int ModuleRtpRtcpImpl::RtxSendStatus() const { + return rtp_sender_ ? rtp_sender_->packet_generator.RtxStatus() : kRtxOff; +} + +void ModuleRtpRtcpImpl::SetRtxSendPayloadType(int payload_type, + int associated_payload_type) { + rtp_sender_->packet_generator.SetRtxPayloadType(payload_type, + associated_payload_type); +} + +absl::optional ModuleRtpRtcpImpl::RtxSsrc() const { + return rtp_sender_ ? rtp_sender_->packet_generator.RtxSsrc() : absl::nullopt; +} + +absl::optional ModuleRtpRtcpImpl::FlexfecSsrc() const { + if (rtp_sender_) { + return rtp_sender_->packet_generator.FlexfecSsrc(); + } + return absl::nullopt; +} + +void ModuleRtpRtcpImpl::IncomingRtcpPacket( + rtc::ArrayView rtcp_packet) { + rtcp_receiver_.IncomingPacket(rtcp_packet); +} + +void ModuleRtpRtcpImpl::RegisterSendPayloadFrequency(int payload_type, + int payload_frequency) { + rtcp_sender_.SetRtpClockRate(payload_type, payload_frequency); +} + +int32_t ModuleRtpRtcpImpl::DeRegisterSendPayload(const int8_t payload_type) { + return 0; +} + +uint32_t ModuleRtpRtcpImpl::StartTimestamp() const { + return rtp_sender_->packet_generator.TimestampOffset(); +} + +// Configure start timestamp, default is a random number. +void ModuleRtpRtcpImpl::SetStartTimestamp(const uint32_t timestamp) { + rtcp_sender_.SetTimestampOffset(timestamp); + rtp_sender_->packet_generator.SetTimestampOffset(timestamp); + rtp_sender_->packet_sender.SetTimestampOffset(timestamp); +} + +uint16_t ModuleRtpRtcpImpl::SequenceNumber() const { + MutexLock lock(&rtp_sender_->sequencer_mutex); + return rtp_sender_->sequencer_.media_sequence_number(); +} + +// Set SequenceNumber, default is a random number. +void ModuleRtpRtcpImpl::SetSequenceNumber(const uint16_t seq_num) { + MutexLock lock(&rtp_sender_->sequencer_mutex); + rtp_sender_->sequencer_.set_media_sequence_number(seq_num); +} + +void ModuleRtpRtcpImpl::SetRtpState(const RtpState& rtp_state) { + MutexLock lock(&rtp_sender_->sequencer_mutex); + rtp_sender_->packet_generator.SetRtpState(rtp_state); + rtp_sender_->sequencer_.SetRtpState(rtp_state); + rtcp_sender_.SetTimestampOffset(rtp_state.start_timestamp); +} + +void ModuleRtpRtcpImpl::SetRtxState(const RtpState& rtp_state) { + MutexLock lock(&rtp_sender_->sequencer_mutex); + rtp_sender_->packet_generator.SetRtxRtpState(rtp_state); + rtp_sender_->sequencer_.set_rtx_sequence_number(rtp_state.sequence_number); +} + +RtpState ModuleRtpRtcpImpl::GetRtpState() const { + MutexLock lock(&rtp_sender_->sequencer_mutex); + RtpState state = rtp_sender_->packet_generator.GetRtpState(); + rtp_sender_->sequencer_.PopulateRtpState(state); + return state; +} + +RtpState ModuleRtpRtcpImpl::GetRtxState() const { + MutexLock lock(&rtp_sender_->sequencer_mutex); + RtpState state = rtp_sender_->packet_generator.GetRtxRtpState(); + state.sequence_number = rtp_sender_->sequencer_.rtx_sequence_number(); + return state; +} + +void ModuleRtpRtcpImpl::SetMid(absl::string_view mid) { + if (rtp_sender_) { + rtp_sender_->packet_generator.SetMid(mid); + } + // TODO(bugs.webrtc.org/4050): If we end up supporting the MID SDES item for + // RTCP, this will need to be passed down to the RTCPSender also. +} + +// TODO(pbos): Handle media and RTX streams separately (separate RTCP +// feedbacks). +RTCPSender::FeedbackState ModuleRtpRtcpImpl::GetFeedbackState() { + RTCPSender::FeedbackState state; + // This is called also when receiver_only is true. Hence below + // checks that rtp_sender_ exists. + if (rtp_sender_) { + StreamDataCounters rtp_stats; + StreamDataCounters rtx_stats; + rtp_sender_->packet_sender.GetDataCounters(&rtp_stats, &rtx_stats); + state.packets_sent = + rtp_stats.transmitted.packets + rtx_stats.transmitted.packets; + state.media_bytes_sent = rtp_stats.transmitted.payload_bytes + + rtx_stats.transmitted.payload_bytes; + state.send_bitrate = rtp_sender_->packet_sender.GetSendRates().Sum(); + } + state.receiver = &rtcp_receiver_; + + if (absl::optional last_sr = + rtcp_receiver_.GetSenderReportStats(); + last_sr.has_value()) { + state.remote_sr = CompactNtp(last_sr->last_remote_timestamp); + state.last_rr = last_sr->last_arrival_timestamp; + } + + state.last_xr_rtis = rtcp_receiver_.ConsumeReceivedXrReferenceTimeInfo(); + + return state; +} + +int32_t ModuleRtpRtcpImpl::SetSendingStatus(const bool sending) { + if (rtcp_sender_.Sending() != sending) { + // Sends RTCP BYE when going from true to false + rtcp_sender_.SetSendingStatus(GetFeedbackState(), sending); + } + return 0; +} + +bool ModuleRtpRtcpImpl::Sending() const { + return rtcp_sender_.Sending(); +} + +void ModuleRtpRtcpImpl::SetSendingMediaStatus(const bool sending) { + rtp_sender_->packet_generator.SetSendingMediaStatus(sending); +} + +bool ModuleRtpRtcpImpl::SendingMedia() const { + return rtp_sender_ ? rtp_sender_->packet_generator.SendingMedia() : false; +} + +bool ModuleRtpRtcpImpl::IsAudioConfigured() const { + return rtp_sender_ ? rtp_sender_->packet_generator.IsAudioConfigured() + : false; +} + +void ModuleRtpRtcpImpl::SetAsPartOfAllocation(bool part_of_allocation) { + RTC_CHECK(rtp_sender_); + rtp_sender_->packet_sender.ForceIncludeSendPacketsInAllocation( + part_of_allocation); +} + +bool ModuleRtpRtcpImpl::OnSendingRtpFrame(uint32_t timestamp, + int64_t capture_time_ms, + int payload_type, + bool force_sender_report) { + if (!Sending()) + return false; + + // TODO(bugs.webrtc.org/12873): Migrate this method and it's users to use + // optional Timestamps. + absl::optional capture_time; + if (capture_time_ms > 0) { + capture_time = Timestamp::Millis(capture_time_ms); + } + absl::optional payload_type_optional; + if (payload_type >= 0) + payload_type_optional = payload_type; + rtcp_sender_.SetLastRtpTime(timestamp, capture_time, payload_type_optional); + // Make sure an RTCP report isn't queued behind a key frame. + if (rtcp_sender_.TimeToSendRTCPReport(force_sender_report)) + rtcp_sender_.SendRTCP(GetFeedbackState(), kRtcpReport); + + return true; +} + +bool ModuleRtpRtcpImpl::TrySendPacket(std::unique_ptr packet, + const PacedPacketInfo& pacing_info) { + RTC_DCHECK(rtp_sender_); + // TODO(sprang): Consider if we can remove this check. + if (!rtp_sender_->packet_generator.SendingMedia()) { + return false; + } + { + MutexLock lock(&rtp_sender_->sequencer_mutex); + if (packet->packet_type() == RtpPacketMediaType::kPadding && + packet->Ssrc() == rtp_sender_->packet_generator.SSRC() && + !rtp_sender_->sequencer_.CanSendPaddingOnMediaSsrc()) { + // New media packet preempted this generated padding packet, discard it. + return false; + } + bool is_flexfec = + packet->packet_type() == RtpPacketMediaType::kForwardErrorCorrection && + packet->Ssrc() == rtp_sender_->packet_generator.FlexfecSsrc(); + if (!is_flexfec) { + rtp_sender_->sequencer_.Sequence(*packet); + } + } + rtp_sender_->packet_sender.SendPacket(packet.get(), pacing_info); + return true; +} + +void ModuleRtpRtcpImpl::SetFecProtectionParams(const FecProtectionParams&, + const FecProtectionParams&) { + // Deferred FEC not supported in deprecated RTP module. +} + +std::vector> +ModuleRtpRtcpImpl::FetchFecPackets() { + // Deferred FEC not supported in deprecated RTP module. + return {}; +} + +void ModuleRtpRtcpImpl::OnAbortedRetransmissions( + rtc::ArrayView sequence_numbers) { + RTC_DCHECK_NOTREACHED() + << "Stream flushing not supported with legacy rtp modules."; +} + +void ModuleRtpRtcpImpl::OnPacketsAcknowledged( + rtc::ArrayView sequence_numbers) { + RTC_DCHECK(rtp_sender_); + rtp_sender_->packet_history.CullAcknowledgedPackets(sequence_numbers); +} + +bool ModuleRtpRtcpImpl::SupportsPadding() const { + RTC_DCHECK(rtp_sender_); + return rtp_sender_->packet_generator.SupportsPadding(); +} + +bool ModuleRtpRtcpImpl::SupportsRtxPayloadPadding() const { + RTC_DCHECK(rtp_sender_); + return rtp_sender_->packet_generator.SupportsRtxPayloadPadding(); +} + +std::vector> +ModuleRtpRtcpImpl::GeneratePadding(size_t target_size_bytes) { + RTC_DCHECK(rtp_sender_); + MutexLock lock(&rtp_sender_->sequencer_mutex); + return rtp_sender_->packet_generator.GeneratePadding( + target_size_bytes, rtp_sender_->packet_sender.MediaHasBeenSent(), + rtp_sender_->sequencer_.CanSendPaddingOnMediaSsrc()); +} + +std::vector +ModuleRtpRtcpImpl::GetSentRtpPacketInfos( + rtc::ArrayView sequence_numbers) const { + RTC_DCHECK(rtp_sender_); + return rtp_sender_->packet_sender.GetSentRtpPacketInfos(sequence_numbers); +} + +size_t ModuleRtpRtcpImpl::ExpectedPerPacketOverhead() const { + if (!rtp_sender_) { + return 0; + } + return rtp_sender_->packet_generator.ExpectedPerPacketOverhead(); +} + +void ModuleRtpRtcpImpl::OnPacketSendingThreadSwitched() {} + +size_t ModuleRtpRtcpImpl::MaxRtpPacketSize() const { + RTC_DCHECK(rtp_sender_); + return rtp_sender_->packet_generator.MaxRtpPacketSize(); +} + +void ModuleRtpRtcpImpl::SetMaxRtpPacketSize(size_t rtp_packet_size) { + RTC_DCHECK_LE(rtp_packet_size, IP_PACKET_SIZE) + << "rtp packet size too large: " << rtp_packet_size; + RTC_DCHECK_GT(rtp_packet_size, packet_overhead_) + << "rtp packet size too small: " << rtp_packet_size; + + rtcp_sender_.SetMaxRtpPacketSize(rtp_packet_size); + if (rtp_sender_) { + rtp_sender_->packet_generator.SetMaxRtpPacketSize(rtp_packet_size); + } +} + +RtcpMode ModuleRtpRtcpImpl::RTCP() const { + return rtcp_sender_.Status(); +} + +// Configure RTCP status i.e on/off. +void ModuleRtpRtcpImpl::SetRTCPStatus(const RtcpMode method) { + rtcp_sender_.SetRTCPStatus(method); +} + +int32_t ModuleRtpRtcpImpl::SetCNAME(absl::string_view c_name) { + return rtcp_sender_.SetCNAME(c_name); +} + +absl::optional ModuleRtpRtcpImpl::LastRtt() const { + absl::optional rtt = rtcp_receiver_.LastRtt(); + if (!rtt.has_value()) { + MutexLock lock(&mutex_rtt_); + if (rtt_ms_ > 0) { + rtt = TimeDelta::Millis(rtt_ms_); + } + } + return rtt; +} + +TimeDelta ModuleRtpRtcpImpl::ExpectedRetransmissionTime() const { + int64_t expected_retransmission_time_ms = rtt_ms(); + if (expected_retransmission_time_ms > 0) { + return TimeDelta::Millis(expected_retransmission_time_ms); + } + // No rtt available (`kRtpRtcpRttProcessTimeMs` not yet passed?), so try to + // poll avg_rtt_ms directly from rtcp receiver. + if (absl::optional rtt = rtcp_receiver_.AverageRtt()) { + return *rtt; + } + return kDefaultExpectedRetransmissionTime; +} + +// Force a send of an RTCP packet. +// Normal SR and RR are triggered via the process function. +int32_t ModuleRtpRtcpImpl::SendRTCP(RTCPPacketType packet_type) { + return rtcp_sender_.SendRTCP(GetFeedbackState(), packet_type); +} + +void ModuleRtpRtcpImpl::GetSendStreamDataCounters( + StreamDataCounters* rtp_counters, + StreamDataCounters* rtx_counters) const { + rtp_sender_->packet_sender.GetDataCounters(rtp_counters, rtx_counters); +} + +// Received RTCP report. +void ModuleRtpRtcpImpl::RemoteRTCPSenderInfo( + uint32_t* packet_count, uint32_t* octet_count, int64_t* ntp_timestamp_ms, + int64_t* remote_ntp_timestamp_ms) const { + return rtcp_receiver_.RemoteRTCPSenderInfo( + packet_count, octet_count, ntp_timestamp_ms, remote_ntp_timestamp_ms); +} + +std::vector ModuleRtpRtcpImpl::GetLatestReportBlockData() + const { + return rtcp_receiver_.GetLatestReportBlockData(); +} + +absl::optional +ModuleRtpRtcpImpl::GetSenderReportStats() const { + return rtcp_receiver_.GetSenderReportStats(); +} + +absl::optional +ModuleRtpRtcpImpl::GetNonSenderRttStats() const { + // This is not implemented for this legacy class. + return absl::nullopt; +} + +// (REMB) Receiver Estimated Max Bitrate. +void ModuleRtpRtcpImpl::SetRemb(int64_t bitrate_bps, + std::vector ssrcs) { + rtcp_sender_.SetRemb(bitrate_bps, std::move(ssrcs)); +} + +void ModuleRtpRtcpImpl::UnsetRemb() { + rtcp_sender_.UnsetRemb(); +} + +void ModuleRtpRtcpImpl::SetExtmapAllowMixed(bool extmap_allow_mixed) { + rtp_sender_->packet_generator.SetExtmapAllowMixed(extmap_allow_mixed); +} + +void ModuleRtpRtcpImpl::RegisterRtpHeaderExtension(absl::string_view uri, + int id) { + bool registered = + rtp_sender_->packet_generator.RegisterRtpHeaderExtension(uri, id); + RTC_CHECK(registered); +} + +void ModuleRtpRtcpImpl::DeregisterSendRtpHeaderExtension( + absl::string_view uri) { + rtp_sender_->packet_generator.DeregisterRtpHeaderExtension(uri); +} + +void ModuleRtpRtcpImpl::SetTmmbn(std::vector bounding_set) { + rtcp_sender_.SetTmmbn(std::move(bounding_set)); +} + +// Send a Negative acknowledgment packet. +int32_t ModuleRtpRtcpImpl::SendNACK(const uint16_t* nack_list, + const uint16_t size) { + uint16_t nack_length = size; + uint16_t start_id = 0; + int64_t now_ms = clock_->TimeInMilliseconds(); + if (TimeToSendFullNackList(now_ms)) { + nack_last_time_sent_full_ms_ = now_ms; + } else { + // Only send extended list. + if (nack_last_seq_number_sent_ == nack_list[size - 1]) { + // Last sequence number is the same, do not send list. + return 0; + } + // Send new sequence numbers. + for (int i = 0; i < size; ++i) { + if (nack_last_seq_number_sent_ == nack_list[i]) { + start_id = i + 1; + break; + } + } + nack_length = size - start_id; + } + + // Our RTCP NACK implementation is limited to kRtcpMaxNackFields sequence + // numbers per RTCP packet. + if (nack_length > kRtcpMaxNackFields) { + nack_length = kRtcpMaxNackFields; + } + nack_last_seq_number_sent_ = nack_list[start_id + nack_length - 1]; + + return rtcp_sender_.SendRTCP(GetFeedbackState(), kRtcpNack, nack_length, + &nack_list[start_id]); +} + +void ModuleRtpRtcpImpl::SendNack( + const std::vector& sequence_numbers) { + rtcp_sender_.SendRTCP(GetFeedbackState(), kRtcpNack, sequence_numbers.size(), + sequence_numbers.data()); +} + +bool ModuleRtpRtcpImpl::TimeToSendFullNackList(int64_t now) const { + // Use RTT from RtcpRttStats class if provided. + int64_t rtt = rtt_ms(); + if (rtt == 0) { + if (absl::optional average_rtt = rtcp_receiver_.AverageRtt()) { + rtt = average_rtt->ms(); + } + } + + const int64_t kStartUpRttMs = 100; + int64_t wait_time = 5 + ((rtt * 3) >> 1); // 5 + RTT * 1.5. + if (rtt == 0) { + wait_time = kStartUpRttMs; + } + + // Send a full NACK list once within every `wait_time`. + return now - nack_last_time_sent_full_ms_ > wait_time; +} + +// Store the sent packets, needed to answer to Negative acknowledgment requests. +void ModuleRtpRtcpImpl::SetStorePacketsStatus(const bool enable, + const uint16_t number_to_store) { + rtp_sender_->packet_history.SetStorePacketsStatus( + enable ? RtpPacketHistory::StorageMode::kStoreAndCull + : RtpPacketHistory::StorageMode::kDisabled, + number_to_store); +} + +bool ModuleRtpRtcpImpl::StorePackets() const { + return rtp_sender_->packet_history.GetStorageMode() != + RtpPacketHistory::StorageMode::kDisabled; +} + +void ModuleRtpRtcpImpl::SendCombinedRtcpPacket( + std::vector> rtcp_packets) { + rtcp_sender_.SendCombinedRtcpPacket(std::move(rtcp_packets)); +} + +int32_t ModuleRtpRtcpImpl::SendLossNotification(uint16_t last_decoded_seq_num, + uint16_t last_received_seq_num, + bool decodability_flag, + bool buffering_allowed) { + return rtcp_sender_.SendLossNotification( + GetFeedbackState(), last_decoded_seq_num, last_received_seq_num, + decodability_flag, buffering_allowed); +} + +void ModuleRtpRtcpImpl::SetRemoteSSRC(const uint32_t ssrc) { + // Inform about the incoming SSRC. + rtcp_sender_.SetRemoteSSRC(ssrc); + rtcp_receiver_.SetRemoteSSRC(ssrc); +} + +void ModuleRtpRtcpImpl::SetLocalSsrc(uint32_t local_ssrc) { + rtcp_receiver_.set_local_media_ssrc(local_ssrc); + rtcp_sender_.SetSsrc(local_ssrc); +} + +RtpSendRates ModuleRtpRtcpImpl::GetSendRates() const { + return rtp_sender_->packet_sender.GetSendRates(); +} + +void ModuleRtpRtcpImpl::OnRequestSendReport() { + SendRTCP(kRtcpSr); +} + +void ModuleRtpRtcpImpl::OnReceivedNack( + const std::vector& nack_sequence_numbers) { + if (!rtp_sender_) + return; + + if (!StorePackets() || nack_sequence_numbers.empty()) { + return; + } + // Use RTT from RtcpRttStats class if provided. + int64_t rtt = rtt_ms(); + if (rtt == 0) { + if (absl::optional average_rtt = rtcp_receiver_.AverageRtt()) { + rtt = average_rtt->ms(); + } + } + rtp_sender_->packet_generator.OnReceivedNack(nack_sequence_numbers, rtt); +} + +void ModuleRtpRtcpImpl::OnReceivedRtcpReportBlocks( + rtc::ArrayView report_blocks) { + if (rtp_sender_) { + uint32_t ssrc = SSRC(); + absl::optional rtx_ssrc; + if (rtp_sender_->packet_generator.RtxStatus() != kRtxOff) { + rtx_ssrc = rtp_sender_->packet_generator.RtxSsrc(); + } + + for (const ReportBlockData& report_block : report_blocks) { + if (ssrc == report_block.source_ssrc()) { + rtp_sender_->packet_generator.OnReceivedAckOnSsrc( + report_block.extended_highest_sequence_number()); + } else if (rtx_ssrc == report_block.source_ssrc()) { + rtp_sender_->packet_generator.OnReceivedAckOnRtxSsrc( + report_block.extended_highest_sequence_number()); + } + } + } +} + +void ModuleRtpRtcpImpl::set_rtt_ms(int64_t rtt_ms) { + { + MutexLock lock(&mutex_rtt_); + rtt_ms_ = rtt_ms; + } + if (rtp_sender_) { + rtp_sender_->packet_history.SetRtt(TimeDelta::Millis(rtt_ms)); + } +} + +int64_t ModuleRtpRtcpImpl::rtt_ms() const { + MutexLock lock(&mutex_rtt_); + return rtt_ms_; +} + +void ModuleRtpRtcpImpl::SetVideoBitrateAllocation( + const VideoBitrateAllocation& bitrate) { + rtcp_sender_.SetVideoBitrateAllocation(bitrate); +} + +RTPSender* ModuleRtpRtcpImpl::RtpSender() { + return rtp_sender_ ? &rtp_sender_->packet_generator : nullptr; +} + +const RTPSender* ModuleRtpRtcpImpl::RtpSender() const { + return rtp_sender_ ? &rtp_sender_->packet_generator : nullptr; +} + +} // namespace webrtc -- cgit v1.2.3