| /* |
| * 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. |
| */ |
| |
| #ifndef MODULES_RTP_RTCP_SOURCE_RTP_RTCP_IMPL2_H_ |
| #define MODULES_RTP_RTCP_SOURCE_RTP_RTCP_IMPL2_H_ |
| |
| #include <stddef.h> |
| #include <stdint.h> |
| |
| #include <memory> |
| #include <set> |
| #include <string> |
| #include <vector> |
| |
| #include "absl/strings/string_view.h" |
| #include "absl/types/optional.h" |
| #include "api/environment/environment.h" |
| #include "api/rtp_headers.h" |
| #include "api/sequence_checker.h" |
| #include "api/task_queue/pending_task_safety_flag.h" |
| #include "api/task_queue/task_queue_base.h" |
| #include "api/units/time_delta.h" |
| #include "api/video/video_bitrate_allocation.h" |
| #include "modules/include/module_fec_types.h" |
| #include "modules/rtp_rtcp/include/rtp_rtcp_defines.h" // RTCPPacketType |
| #include "modules/rtp_rtcp/source/packet_sequencer.h" |
| #include "modules/rtp_rtcp/source/rtcp_packet/tmmb_item.h" |
| #include "modules/rtp_rtcp/source/rtcp_receiver.h" |
| #include "modules/rtp_rtcp/source/rtcp_sender.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_sender.h" |
| #include "modules/rtp_rtcp/source/rtp_sender_egress.h" |
| #include "rtc_base/gtest_prod_util.h" |
| #include "rtc_base/synchronization/mutex.h" |
| #include "rtc_base/system/no_unique_address.h" |
| #include "rtc_base/task_utils/repeating_task.h" |
| #include "rtc_base/thread_annotations.h" |
| |
| namespace webrtc { |
| |
| class Clock; |
| struct PacedPacketInfo; |
| struct RTPVideoHeader; |
| |
| class ModuleRtpRtcpImpl2 final : public RtpRtcpInterface, |
| public RTCPReceiver::ModuleRtpRtcp { |
| public: |
| ModuleRtpRtcpImpl2(const Environment& env, |
| const RtpRtcpInterface::Configuration& configuration); |
| ~ModuleRtpRtcpImpl2() override; |
| |
| // This method is provided to easy with migrating away from the |
| // RtpRtcp::Create factory method. Since this is an internal implementation |
| // detail though, creating an instance of ModuleRtpRtcpImpl2 directly should |
| // be fine. |
| static std::unique_ptr<ModuleRtpRtcpImpl2> Create( |
| const Configuration& configuration); |
| |
| // Receiver part. |
| |
| // Called when we receive an RTCP packet. |
| void IncomingRtcpPacket( |
| rtc::ArrayView<const uint8_t> incoming_packet) override; |
| |
| void SetRemoteSSRC(uint32_t ssrc) override; |
| |
| void SetLocalSsrc(uint32_t local_ssrc) override; |
| |
| // Sender part. |
| void RegisterSendPayloadFrequency(int payload_type, |
| int payload_frequency) override; |
| |
| int32_t DeRegisterSendPayload(int8_t payload_type) override; |
| |
| void SetExtmapAllowMixed(bool extmap_allow_mixed) override; |
| |
| void RegisterRtpHeaderExtension(absl::string_view uri, int id) override; |
| void DeregisterSendRtpHeaderExtension(absl::string_view uri) override; |
| |
| bool SupportsPadding() const override; |
| bool SupportsRtxPayloadPadding() const override; |
| |
| // Get start timestamp. |
| uint32_t StartTimestamp() const override; |
| |
| // Configure start timestamp, default is a random number. |
| void SetStartTimestamp(uint32_t timestamp) override; |
| |
| uint16_t SequenceNumber() const override; |
| |
| // Set SequenceNumber, default is a random number. |
| void SetSequenceNumber(uint16_t seq) override; |
| |
| void SetRtpState(const RtpState& rtp_state) override; |
| void SetRtxState(const RtpState& rtp_state) override; |
| RtpState GetRtpState() const override; |
| RtpState GetRtxState() const override; |
| |
| void SetNonSenderRttMeasurement(bool enabled) override; |
| |
| uint32_t SSRC() const override { return rtcp_sender_.SSRC(); } |
| |
| // Semantically identical to `SSRC()` but must be called on the packet |
| // delivery thread/tq and returns the ssrc that maps to |
| // RtpRtcpInterface::Configuration::local_media_ssrc. |
| uint32_t local_media_ssrc() const; |
| |
| void SetMid(absl::string_view mid) override; |
| |
| RTCPSender::FeedbackState GetFeedbackState(); |
| |
| void SetRtxSendStatus(int mode) override; |
| int RtxSendStatus() const override; |
| absl::optional<uint32_t> RtxSsrc() const override; |
| |
| void SetRtxSendPayloadType(int payload_type, |
| int associated_payload_type) override; |
| |
| absl::optional<uint32_t> FlexfecSsrc() const override; |
| |
| // Sends kRtcpByeCode when going from true to false. |
| int32_t SetSendingStatus(bool sending) override; |
| |
| bool Sending() const override; |
| |
| // Drops or relays media packets. |
| void SetSendingMediaStatus(bool sending) override; |
| |
| bool SendingMedia() const override; |
| |
| bool IsAudioConfigured() const override; |
| |
| void SetAsPartOfAllocation(bool part_of_allocation) override; |
| |
| bool OnSendingRtpFrame(uint32_t timestamp, |
| int64_t capture_time_ms, |
| int payload_type, |
| bool force_sender_report) override; |
| |
| bool CanSendPacket(const RtpPacketToSend& packet) const override; |
| |
| void AssignSequenceNumber(RtpPacketToSend& packet) override; |
| |
| void SendPacket(std::unique_ptr<RtpPacketToSend> packet, |
| const PacedPacketInfo& pacing_info) override; |
| |
| bool TrySendPacket(std::unique_ptr<RtpPacketToSend> packet, |
| const PacedPacketInfo& pacing_info) override; |
| void OnBatchComplete() override; |
| |
| void SetFecProtectionParams(const FecProtectionParams& delta_params, |
| const FecProtectionParams& key_params) override; |
| |
| std::vector<std::unique_ptr<RtpPacketToSend>> FetchFecPackets() override; |
| |
| void OnAbortedRetransmissions( |
| rtc::ArrayView<const uint16_t> sequence_numbers) override; |
| |
| void OnPacketsAcknowledged( |
| rtc::ArrayView<const uint16_t> sequence_numbers) override; |
| |
| std::vector<std::unique_ptr<RtpPacketToSend>> GeneratePadding( |
| size_t target_size_bytes) override; |
| |
| std::vector<RtpSequenceNumberMap::Info> GetSentRtpPacketInfos( |
| rtc::ArrayView<const uint16_t> sequence_numbers) const override; |
| |
| size_t ExpectedPerPacketOverhead() const override; |
| |
| void OnPacketSendingThreadSwitched() override; |
| |
| // RTCP part. |
| |
| // Get RTCP status. |
| RtcpMode RTCP() const override; |
| |
| // Configure RTCP status i.e on/off. |
| void SetRTCPStatus(RtcpMode method) override; |
| |
| // Set RTCP CName. |
| int32_t SetCNAME(absl::string_view c_name) override; |
| |
| // Get RoundTripTime. |
| absl::optional<TimeDelta> LastRtt() const override; |
| |
| TimeDelta ExpectedRetransmissionTime() const override; |
| |
| // Force a send of an RTCP packet. |
| // Normal SR and RR are triggered via the task queue that's current when this |
| // object is created. |
| int32_t SendRTCP(RTCPPacketType rtcpPacketType) override; |
| |
| void GetSendStreamDataCounters( |
| StreamDataCounters* rtp_counters, |
| StreamDataCounters* rtx_counters) const override; |
| |
| // A snapshot of the most recent Report Block with additional data of |
| // interest to statistics. Used to implement RTCRemoteInboundRtpStreamStats. |
| // Within this list, the `ReportBlockData::source_ssrc()`, which is the SSRC |
| // of the corresponding outbound RTP stream, is unique. |
| std::vector<ReportBlockData> GetLatestReportBlockData() const override; |
| absl::optional<SenderReportStats> GetSenderReportStats() const override; |
| absl::optional<NonSenderRttStats> GetNonSenderRttStats() const override; |
| |
| // (REMB) Receiver Estimated Max Bitrate. |
| void SetRemb(int64_t bitrate_bps, std::vector<uint32_t> ssrcs) override; |
| void UnsetRemb() override; |
| |
| void SetTmmbn(std::vector<rtcp::TmmbItem> bounding_set) override; |
| |
| size_t MaxRtpPacketSize() const override; |
| |
| void SetMaxRtpPacketSize(size_t max_packet_size) override; |
| |
| // (NACK) Negative acknowledgment part. |
| |
| // Send a Negative acknowledgment packet. |
| // TODO(philipel): Deprecate SendNACK and use SendNack instead. |
| int32_t SendNACK(const uint16_t* nack_list, uint16_t size) override; |
| |
| void SendNack(const std::vector<uint16_t>& sequence_numbers) override; |
| |
| // Store the sent packets, needed to answer to a negative acknowledgment |
| // requests. |
| void SetStorePacketsStatus(bool enable, uint16_t number_to_store) override; |
| |
| void SendCombinedRtcpPacket( |
| std::vector<std::unique_ptr<rtcp::RtcpPacket>> rtcp_packets) override; |
| |
| // Video part. |
| int32_t SendLossNotification(uint16_t last_decoded_seq_num, |
| uint16_t last_received_seq_num, |
| bool decodability_flag, |
| bool buffering_allowed) override; |
| |
| RtpSendRates GetSendRates() const override; |
| |
| void OnReceivedNack( |
| const std::vector<uint16_t>& nack_sequence_numbers) override; |
| void OnReceivedRtcpReportBlocks( |
| rtc::ArrayView<const ReportBlockData> report_blocks) override; |
| void OnRequestSendReport() override; |
| |
| void SetVideoBitrateAllocation( |
| const VideoBitrateAllocation& bitrate) override; |
| |
| RTPSender* RtpSender() override; |
| const RTPSender* RtpSender() const override; |
| |
| private: |
| FRIEND_TEST_ALL_PREFIXES(RtpRtcpImpl2Test, Rtt); |
| FRIEND_TEST_ALL_PREFIXES(RtpRtcpImpl2Test, RttForReceiverOnly); |
| |
| struct RtpSenderContext { |
| explicit RtpSenderContext(TaskQueueBase& worker_queue, |
| const RtpRtcpInterface::Configuration& config); |
| // Storage of packets, for retransmissions and padding, if applicable. |
| RtpPacketHistory packet_history; |
| SequenceChecker sequencing_checker; |
| // Handles sequence number assignment and padding timestamp generation. |
| PacketSequencer sequencer RTC_GUARDED_BY(sequencing_checker); |
| // Handles final time timestamping/stats/etc and handover to Transport. |
| RtpSenderEgress packet_sender; |
| // If no paced sender configured, this class will be used to pass packets |
| // from `packet_generator_` to `packet_sender_`. |
| RtpSenderEgress::NonPacedPacketSender non_paced_sender; |
| // Handles creation of RTP packets to be sent. |
| RTPSender packet_generator; |
| }; |
| |
| explicit ModuleRtpRtcpImpl2(const Configuration& configuration); |
| |
| void set_rtt_ms(int64_t rtt_ms); |
| int64_t rtt_ms() const; |
| |
| bool TimeToSendFullNackList(int64_t now) const; |
| |
| // Called on a timer, once a second, on the worker_queue_, to update the RTT, |
| // check if we need to send RTCP report, send TMMBR updates and fire events. |
| void PeriodicUpdate(); |
| |
| // Returns true if the module is configured to store packets. |
| bool StorePackets() const; |
| |
| // Used from RtcpSenderMediator to maybe send rtcp. |
| void MaybeSendRtcp() RTC_RUN_ON(worker_queue_); |
| |
| // Called when `rtcp_sender_` informs of the next RTCP instant. The method may |
| // be called on various sequences, and is called under a RTCPSenderLock. |
| void ScheduleRtcpSendEvaluation(TimeDelta duration); |
| |
| // Helper method combating too early delayed calls from task queues. |
| // TODO(bugs.webrtc.org/12889): Consider removing this function when the issue |
| // is resolved. |
| void MaybeSendRtcpAtOrAfterTimestamp(Timestamp execution_time) |
| RTC_RUN_ON(worker_queue_); |
| |
| // Schedules a call to MaybeSendRtcpAtOrAfterTimestamp delayed by `duration`. |
| void ScheduleMaybeSendRtcpAtOrAfterTimestamp(Timestamp execution_time, |
| TimeDelta duration); |
| |
| TaskQueueBase* const worker_queue_; |
| RTC_NO_UNIQUE_ADDRESS SequenceChecker rtcp_thread_checker_; |
| |
| std::unique_ptr<RtpSenderContext> rtp_sender_; |
| RTCPSender rtcp_sender_; |
| RTCPReceiver rtcp_receiver_; |
| |
| Clock* const clock_; |
| |
| uint16_t packet_overhead_; |
| |
| // Send side |
| int64_t nack_last_time_sent_full_ms_; |
| uint16_t nack_last_seq_number_sent_; |
| |
| RtcpRttStats* const rtt_stats_; |
| RepeatingTaskHandle rtt_update_task_ RTC_GUARDED_BY(worker_queue_); |
| |
| // The processed RTT from RtcpRttStats. |
| mutable Mutex mutex_rtt_; |
| int64_t rtt_ms_ RTC_GUARDED_BY(mutex_rtt_); |
| |
| RTC_NO_UNIQUE_ADDRESS ScopedTaskSafety task_safety_; |
| }; |
| |
| } // namespace webrtc |
| |
| #endif // MODULES_RTP_RTCP_SOURCE_RTP_RTCP_IMPL2_H_ |