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author | Daniel Baumann <daniel.baumann@progress-linux.org> | 2024-04-19 00:47:55 +0000 |
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committer | Daniel Baumann <daniel.baumann@progress-linux.org> | 2024-04-19 00:47:55 +0000 |
commit | 26a029d407be480d791972afb5975cf62c9360a6 (patch) | |
tree | f435a8308119effd964b339f76abb83a57c29483 /third_party/libwebrtc/modules/rtp_rtcp/source/deprecated | |
parent | Initial commit. (diff) | |
download | firefox-26a029d407be480d791972afb5975cf62c9360a6.tar.xz firefox-26a029d407be480d791972afb5975cf62c9360a6.zip |
Adding upstream version 124.0.1.upstream/124.0.1
Signed-off-by: Daniel Baumann <daniel.baumann@progress-linux.org>
Diffstat (limited to 'third_party/libwebrtc/modules/rtp_rtcp/source/deprecated')
2 files changed, 511 insertions, 0 deletions
diff --git a/third_party/libwebrtc/modules/rtp_rtcp/source/deprecated/deprecated_rtp_sender_egress.cc b/third_party/libwebrtc/modules/rtp_rtcp/source/deprecated/deprecated_rtp_sender_egress.cc new file mode 100644 index 0000000000..59ff9d70f1 --- /dev/null +++ b/third_party/libwebrtc/modules/rtp_rtcp/source/deprecated/deprecated_rtp_sender_egress.cc @@ -0,0 +1,374 @@ +/* + * Copyright (c) 2019 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/deprecated/deprecated_rtp_sender_egress.h" + +#include <limits> +#include <memory> +#include <utility> + +#include "absl/strings/match.h" +#include "api/units/timestamp.h" +#include "logging/rtc_event_log/events/rtc_event_rtp_packet_outgoing.h" +#include "modules/remote_bitrate_estimator/test/bwe_test_logging.h" +#include "rtc_base/logging.h" + +namespace webrtc { +namespace { +constexpr uint32_t kTimestampTicksPerMs = 90; +constexpr TimeDelta kBitrateStatisticsWindow = TimeDelta::Seconds(1); +constexpr size_t kRtpSequenceNumberMapMaxEntries = 1 << 13; + +} // namespace + +DEPRECATED_RtpSenderEgress::NonPacedPacketSender::NonPacedPacketSender( + DEPRECATED_RtpSenderEgress* sender, + PacketSequencer* sequence_number_assigner) + : transport_sequence_number_(0), + sender_(sender), + sequence_number_assigner_(sequence_number_assigner) { + RTC_DCHECK(sequence_number_assigner_); +} +DEPRECATED_RtpSenderEgress::NonPacedPacketSender::~NonPacedPacketSender() = + default; + +void DEPRECATED_RtpSenderEgress::NonPacedPacketSender::EnqueuePackets( + std::vector<std::unique_ptr<RtpPacketToSend>> packets) { + for (auto& packet : packets) { + // Assign sequence numbers, but not for flexfec which is already running on + // an internally maintained sequence number series. + if (packet->Ssrc() != sender_->FlexFecSsrc()) { + sequence_number_assigner_->Sequence(*packet); + } + if (!packet->SetExtension<TransportSequenceNumber>( + ++transport_sequence_number_)) { + --transport_sequence_number_; + } + packet->ReserveExtension<TransmissionOffset>(); + packet->ReserveExtension<AbsoluteSendTime>(); + sender_->SendPacket(packet.get(), PacedPacketInfo()); + } +} + +DEPRECATED_RtpSenderEgress::DEPRECATED_RtpSenderEgress( + const RtpRtcpInterface::Configuration& config, + RtpPacketHistory* packet_history) + : ssrc_(config.local_media_ssrc), + rtx_ssrc_(config.rtx_send_ssrc), + flexfec_ssrc_(config.fec_generator ? config.fec_generator->FecSsrc() + : absl::nullopt), + populate_network2_timestamp_(config.populate_network2_timestamp), + clock_(config.clock), + packet_history_(packet_history), + transport_(config.outgoing_transport), + event_log_(config.event_log), + is_audio_(config.audio), + need_rtp_packet_infos_(config.need_rtp_packet_infos), + transport_feedback_observer_(config.transport_feedback_callback), + send_packet_observer_(config.send_packet_observer), + rtp_stats_callback_(config.rtp_stats_callback), + bitrate_callback_(config.send_bitrate_observer), + media_has_been_sent_(false), + force_part_of_allocation_(false), + timestamp_offset_(0), + send_rates_(kNumMediaTypes, BitrateTracker(kBitrateStatisticsWindow)), + rtp_sequence_number_map_(need_rtp_packet_infos_ + ? std::make_unique<RtpSequenceNumberMap>( + kRtpSequenceNumberMapMaxEntries) + : nullptr) {} + +void DEPRECATED_RtpSenderEgress::SendPacket( + RtpPacketToSend* packet, + const PacedPacketInfo& pacing_info) { + RTC_DCHECK(packet); + + const uint32_t packet_ssrc = packet->Ssrc(); + RTC_DCHECK(packet->packet_type().has_value()); + RTC_DCHECK(HasCorrectSsrc(*packet)); + Timestamp now = clock_->CurrentTime(); + int64_t now_ms = now.ms(); + + if (is_audio_) { +#if BWE_TEST_LOGGING_COMPILE_TIME_ENABLE + BWE_TEST_LOGGING_PLOT_WITH_SSRC(1, "AudioTotBitrate_kbps", now_ms, + GetSendRates().Sum().kbps(), packet_ssrc); + BWE_TEST_LOGGING_PLOT_WITH_SSRC( + 1, "AudioNackBitrate_kbps", now_ms, + GetSendRates()[RtpPacketMediaType::kRetransmission].kbps(), + packet_ssrc); +#endif + } else { +#if BWE_TEST_LOGGING_COMPILE_TIME_ENABLE + BWE_TEST_LOGGING_PLOT_WITH_SSRC(1, "VideoTotBitrate_kbps", now_ms, + GetSendRates().Sum().kbps(), packet_ssrc); + BWE_TEST_LOGGING_PLOT_WITH_SSRC( + 1, "VideoNackBitrate_kbps", now_ms, + GetSendRates()[RtpPacketMediaType::kRetransmission].kbps(), + packet_ssrc); +#endif + } + + PacketOptions options; + { + MutexLock lock(&lock_); + options.included_in_allocation = force_part_of_allocation_; + + if (need_rtp_packet_infos_ && + packet->packet_type() == RtpPacketToSend::Type::kVideo) { + RTC_DCHECK(rtp_sequence_number_map_); + // Last packet of a frame, add it to sequence number info map. + const uint32_t timestamp = packet->Timestamp() - timestamp_offset_; + bool is_first_packet_of_frame = packet->is_first_packet_of_frame(); + bool is_last_packet_of_frame = packet->Marker(); + + rtp_sequence_number_map_->InsertPacket( + packet->SequenceNumber(), + RtpSequenceNumberMap::Info(timestamp, is_first_packet_of_frame, + is_last_packet_of_frame)); + } + } + + // Bug webrtc:7859. While FEC is invoked from rtp_sender_video, and not after + // the pacer, these modifications of the header below are happening after the + // FEC protection packets are calculated. This will corrupt recovered packets + // at the same place. It's not an issue for extensions, which are present in + // all the packets (their content just may be incorrect on recovered packets). + // In case of VideoTimingExtension, since it's present not in every packet, + // data after rtp header may be corrupted if these packets are protected by + // the FEC. + int64_t diff_ms = now_ms - packet->capture_time().ms(); + if (packet->HasExtension<TransmissionOffset>()) { + packet->SetExtension<TransmissionOffset>(kTimestampTicksPerMs * diff_ms); + } + if (packet->HasExtension<AbsoluteSendTime>()) { + packet->SetExtension<AbsoluteSendTime>(AbsoluteSendTime::To24Bits(now)); + } + + if (packet->HasExtension<VideoTimingExtension>()) { + if (populate_network2_timestamp_) { + packet->set_network2_time(now); + } else { + packet->set_pacer_exit_time(now); + } + } + + const bool is_media = packet->packet_type() == RtpPacketMediaType::kAudio || + packet->packet_type() == RtpPacketMediaType::kVideo; + + // Downstream code actually uses this flag to distinguish between media and + // everything else. + options.is_retransmit = !is_media; + if (auto packet_id = packet->GetExtension<TransportSequenceNumber>()) { + options.packet_id = *packet_id; + options.included_in_feedback = true; + options.included_in_allocation = true; + AddPacketToTransportFeedback(*packet_id, *packet, pacing_info); + } + + options.additional_data = packet->additional_data(); + + if (packet->packet_type() != RtpPacketMediaType::kPadding && + packet->packet_type() != RtpPacketMediaType::kRetransmission) { + UpdateOnSendPacket(options.packet_id, packet->capture_time().ms(), + packet_ssrc); + } + + const bool send_success = SendPacketToNetwork(*packet, options, pacing_info); + + // Put packet in retransmission history or update pending status even if + // actual sending fails. + if (is_media && packet->allow_retransmission()) { + packet_history_->PutRtpPacket(std::make_unique<RtpPacketToSend>(*packet), + now); + } else if (packet->retransmitted_sequence_number()) { + packet_history_->MarkPacketAsSent(*packet->retransmitted_sequence_number()); + } + + if (send_success) { + MutexLock lock(&lock_); + UpdateRtpStats(*packet); + media_has_been_sent_ = true; + } +} + +void DEPRECATED_RtpSenderEgress::ProcessBitrateAndNotifyObservers() { + if (!bitrate_callback_) + return; + + MutexLock lock(&lock_); + RtpSendRates send_rates = GetSendRatesLocked(); + bitrate_callback_->Notify( + send_rates.Sum().bps(), + send_rates[RtpPacketMediaType::kRetransmission].bps(), ssrc_); +} + +RtpSendRates DEPRECATED_RtpSenderEgress::GetSendRates() const { + MutexLock lock(&lock_); + return GetSendRatesLocked(); +} + +RtpSendRates DEPRECATED_RtpSenderEgress::GetSendRatesLocked() const { + const Timestamp now = clock_->CurrentTime(); + RtpSendRates current_rates; + for (size_t i = 0; i < kNumMediaTypes; ++i) { + RtpPacketMediaType type = static_cast<RtpPacketMediaType>(i); + current_rates[type] = send_rates_[i].Rate(now).value_or(DataRate::Zero()); + } + return current_rates; +} + +void DEPRECATED_RtpSenderEgress::GetDataCounters( + StreamDataCounters* rtp_stats, + StreamDataCounters* rtx_stats) const { + MutexLock lock(&lock_); + *rtp_stats = rtp_stats_; + *rtx_stats = rtx_rtp_stats_; +} + +void DEPRECATED_RtpSenderEgress::ForceIncludeSendPacketsInAllocation( + bool part_of_allocation) { + MutexLock lock(&lock_); + force_part_of_allocation_ = part_of_allocation; +} + +bool DEPRECATED_RtpSenderEgress::MediaHasBeenSent() const { + MutexLock lock(&lock_); + return media_has_been_sent_; +} + +void DEPRECATED_RtpSenderEgress::SetMediaHasBeenSent(bool media_sent) { + MutexLock lock(&lock_); + media_has_been_sent_ = media_sent; +} + +void DEPRECATED_RtpSenderEgress::SetTimestampOffset(uint32_t timestamp) { + MutexLock lock(&lock_); + timestamp_offset_ = timestamp; +} + +std::vector<RtpSequenceNumberMap::Info> +DEPRECATED_RtpSenderEgress::GetSentRtpPacketInfos( + rtc::ArrayView<const uint16_t> sequence_numbers) const { + RTC_DCHECK(!sequence_numbers.empty()); + if (!need_rtp_packet_infos_) { + return std::vector<RtpSequenceNumberMap::Info>(); + } + + std::vector<RtpSequenceNumberMap::Info> results; + results.reserve(sequence_numbers.size()); + + MutexLock lock(&lock_); + for (uint16_t sequence_number : sequence_numbers) { + const auto& info = rtp_sequence_number_map_->Get(sequence_number); + if (!info) { + // The empty vector will be returned. We can delay the clearing + // of the vector until after we exit the critical section. + return std::vector<RtpSequenceNumberMap::Info>(); + } + results.push_back(*info); + } + + return results; +} + +bool DEPRECATED_RtpSenderEgress::HasCorrectSsrc( + const RtpPacketToSend& packet) const { + switch (*packet.packet_type()) { + case RtpPacketMediaType::kAudio: + case RtpPacketMediaType::kVideo: + return packet.Ssrc() == ssrc_; + case RtpPacketMediaType::kRetransmission: + case RtpPacketMediaType::kPadding: + // Both padding and retransmission must be on either the media or the + // RTX stream. + return packet.Ssrc() == rtx_ssrc_ || packet.Ssrc() == ssrc_; + case RtpPacketMediaType::kForwardErrorCorrection: + // FlexFEC is on separate SSRC, ULPFEC uses media SSRC. + return packet.Ssrc() == ssrc_ || packet.Ssrc() == flexfec_ssrc_; + } + return false; +} + +void DEPRECATED_RtpSenderEgress::AddPacketToTransportFeedback( + uint16_t packet_id, + const RtpPacketToSend& packet, + const PacedPacketInfo& pacing_info) { + if (transport_feedback_observer_) { + RtpPacketSendInfo packet_info; + packet_info.media_ssrc = ssrc_; + packet_info.transport_sequence_number = packet_id; + packet_info.rtp_sequence_number = packet.SequenceNumber(); + packet_info.length = packet.size(); + packet_info.pacing_info = pacing_info; + packet_info.packet_type = packet.packet_type(); + transport_feedback_observer_->OnAddPacket(packet_info); + } +} + +void DEPRECATED_RtpSenderEgress::UpdateOnSendPacket(int packet_id, + int64_t capture_time_ms, + uint32_t ssrc) { + if (!send_packet_observer_ || capture_time_ms <= 0 || packet_id == -1) { + return; + } + + send_packet_observer_->OnSendPacket(packet_id, + Timestamp::Millis(capture_time_ms), ssrc); +} + +bool DEPRECATED_RtpSenderEgress::SendPacketToNetwork( + const RtpPacketToSend& packet, + const PacketOptions& options, + const PacedPacketInfo& pacing_info) { + int bytes_sent = -1; + if (transport_) { + bytes_sent = transport_->SendRtp(packet, options) + ? static_cast<int>(packet.size()) + : -1; + if (event_log_ && bytes_sent > 0) { + event_log_->Log(std::make_unique<RtcEventRtpPacketOutgoing>( + packet, pacing_info.probe_cluster_id)); + } + } + + if (bytes_sent <= 0) { + RTC_LOG(LS_WARNING) << "Transport failed to send packet."; + return false; + } + return true; +} + +void DEPRECATED_RtpSenderEgress::UpdateRtpStats(const RtpPacketToSend& packet) { + Timestamp now = clock_->CurrentTime(); + + StreamDataCounters* counters = + packet.Ssrc() == rtx_ssrc_ ? &rtx_rtp_stats_ : &rtp_stats_; + + counters->MaybeSetFirstPacketTime(now); + + if (packet.packet_type() == RtpPacketMediaType::kForwardErrorCorrection) { + counters->fec.AddPacket(packet); + } + + if (packet.packet_type() == RtpPacketMediaType::kRetransmission) { + counters->retransmitted.AddPacket(packet); + } + counters->transmitted.AddPacket(packet); + + RTC_DCHECK(packet.packet_type().has_value()); + send_rates_[static_cast<size_t>(*packet.packet_type())].Update(packet.size(), + now); + + if (rtp_stats_callback_) { + rtp_stats_callback_->DataCountersUpdated(*counters, packet.Ssrc()); + } +} + +} // namespace webrtc diff --git a/third_party/libwebrtc/modules/rtp_rtcp/source/deprecated/deprecated_rtp_sender_egress.h b/third_party/libwebrtc/modules/rtp_rtcp/source/deprecated/deprecated_rtp_sender_egress.h new file mode 100644 index 0000000000..9d343c2d08 --- /dev/null +++ b/third_party/libwebrtc/modules/rtp_rtcp/source/deprecated/deprecated_rtp_sender_egress.h @@ -0,0 +1,137 @@ +/* + * Copyright (c) 2019 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. + */ + +#ifndef MODULES_RTP_RTCP_SOURCE_DEPRECATED_DEPRECATED_RTP_SENDER_EGRESS_H_ +#define MODULES_RTP_RTCP_SOURCE_DEPRECATED_DEPRECATED_RTP_SENDER_EGRESS_H_ + +#include <map> +#include <memory> +#include <vector> + +#include "absl/types/optional.h" +#include "api/call/transport.h" +#include "api/rtc_event_log/rtc_event_log.h" +#include "api/units/data_rate.h" +#include "modules/rtp_rtcp/include/rtp_rtcp_defines.h" +#include "modules/rtp_rtcp/source/packet_sequencer.h" +#include "modules/rtp_rtcp/source/rtp_packet_history.h" +#include "modules/rtp_rtcp/source/rtp_packet_to_send.h" +#include "modules/rtp_rtcp/source/rtp_rtcp_interface.h" +#include "modules/rtp_rtcp/source/rtp_sequence_number_map.h" +#include "rtc_base/bitrate_tracker.h" +#include "rtc_base/synchronization/mutex.h" +#include "rtc_base/thread_annotations.h" + +namespace webrtc { + +class DEPRECATED_RtpSenderEgress { + public: + // Helper class that redirects packets directly to the send part of this class + // without passing through an actual paced sender. + class NonPacedPacketSender : public RtpPacketSender { + public: + NonPacedPacketSender(DEPRECATED_RtpSenderEgress* sender, + PacketSequencer* sequence_number_assigner); + virtual ~NonPacedPacketSender(); + + void EnqueuePackets( + std::vector<std::unique_ptr<RtpPacketToSend>> packets) override; + void RemovePacketsForSsrc(uint32_t ssrc) override {} + + private: + uint16_t transport_sequence_number_; + DEPRECATED_RtpSenderEgress* const sender_; + PacketSequencer* sequence_number_assigner_; + }; + + DEPRECATED_RtpSenderEgress(const RtpRtcpInterface::Configuration& config, + RtpPacketHistory* packet_history); + ~DEPRECATED_RtpSenderEgress() = default; + + void SendPacket(RtpPacketToSend* packet, const PacedPacketInfo& pacing_info) + RTC_LOCKS_EXCLUDED(lock_); + uint32_t Ssrc() const { return ssrc_; } + absl::optional<uint32_t> RtxSsrc() const { return rtx_ssrc_; } + absl::optional<uint32_t> FlexFecSsrc() const { return flexfec_ssrc_; } + + void ProcessBitrateAndNotifyObservers() RTC_LOCKS_EXCLUDED(lock_); + RtpSendRates GetSendRates() const RTC_LOCKS_EXCLUDED(lock_); + void GetDataCounters(StreamDataCounters* rtp_stats, + StreamDataCounters* rtx_stats) const + RTC_LOCKS_EXCLUDED(lock_); + + void ForceIncludeSendPacketsInAllocation(bool part_of_allocation) + RTC_LOCKS_EXCLUDED(lock_); + bool MediaHasBeenSent() const RTC_LOCKS_EXCLUDED(lock_); + void SetMediaHasBeenSent(bool media_sent) RTC_LOCKS_EXCLUDED(lock_); + void SetTimestampOffset(uint32_t timestamp) RTC_LOCKS_EXCLUDED(lock_); + + // For each sequence number in `sequence_number`, recall the last RTP packet + // which bore it - its timestamp and whether it was the first and/or last + // packet in that frame. If all of the given sequence numbers could be + // recalled, return a vector with all of them (in corresponding order). + // If any could not be recalled, return an empty vector. + std::vector<RtpSequenceNumberMap::Info> GetSentRtpPacketInfos( + rtc::ArrayView<const uint16_t> sequence_numbers) const + RTC_LOCKS_EXCLUDED(lock_); + + private: + RtpSendRates GetSendRatesLocked() const RTC_EXCLUSIVE_LOCKS_REQUIRED(lock_); + bool HasCorrectSsrc(const RtpPacketToSend& packet) const; + void AddPacketToTransportFeedback(uint16_t packet_id, + const RtpPacketToSend& packet, + const PacedPacketInfo& pacing_info); + void UpdateOnSendPacket(int packet_id, + int64_t capture_time_ms, + uint32_t ssrc); + // Sends packet on to `transport_`, leaving the RTP module. + bool SendPacketToNetwork(const RtpPacketToSend& packet, + const PacketOptions& options, + const PacedPacketInfo& pacing_info); + void UpdateRtpStats(const RtpPacketToSend& packet) + RTC_EXCLUSIVE_LOCKS_REQUIRED(lock_); + + const uint32_t ssrc_; + const absl::optional<uint32_t> rtx_ssrc_; + const absl::optional<uint32_t> flexfec_ssrc_; + const bool populate_network2_timestamp_; + Clock* const clock_; + RtpPacketHistory* const packet_history_; + Transport* const transport_; + RtcEventLog* const event_log_; + const bool is_audio_; + const bool need_rtp_packet_infos_; + + TransportFeedbackObserver* const transport_feedback_observer_; + SendPacketObserver* const send_packet_observer_; + StreamDataCountersCallback* const rtp_stats_callback_; + BitrateStatisticsObserver* const bitrate_callback_; + + mutable Mutex lock_; + bool media_has_been_sent_ RTC_GUARDED_BY(lock_); + bool force_part_of_allocation_ RTC_GUARDED_BY(lock_); + uint32_t timestamp_offset_ RTC_GUARDED_BY(lock_); + + StreamDataCounters rtp_stats_ RTC_GUARDED_BY(lock_); + StreamDataCounters rtx_rtp_stats_ RTC_GUARDED_BY(lock_); + // One element per value in RtpPacketMediaType, with index matching value. + std::vector<BitrateTracker> send_rates_ RTC_GUARDED_BY(lock_); + + // Maps sent packets' sequence numbers to a tuple consisting of: + // 1. The timestamp, without the randomizing offset mandated by the RFC. + // 2. Whether the packet was the first in its frame. + // 3. Whether the packet was the last in its frame. + const std::unique_ptr<RtpSequenceNumberMap> rtp_sequence_number_map_ + RTC_GUARDED_BY(lock_); +}; + +} // namespace webrtc + +#endif // MODULES_RTP_RTCP_SOURCE_DEPRECATED_DEPRECATED_RTP_SENDER_EGRESS_H_ |