/* * 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_IMPL_H_ #define MODULES_RTP_RTCP_SOURCE_RTP_RTCP_IMPL_H_ #include #include #include #include #include #include #include "absl/types/optional.h" #include "api/rtp_headers.h" #include "api/video/video_bitrate_allocation.h" #include "modules/include/module_fec_types.h" #include "modules/remote_bitrate_estimator/include/remote_bitrate_estimator.h" #include "modules/rtp_rtcp/include/rtp_rtcp.h" #include "modules/rtp_rtcp/include/rtp_rtcp_defines.h" // RTCPPacketType #include "modules/rtp_rtcp/source/deprecated/deprecated_rtp_sender_egress.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 "rtc_base/gtest_prod_util.h" #include "rtc_base/synchronization/mutex.h" namespace webrtc { class Clock; struct PacedPacketInfo; struct RTPVideoHeader; // DEPRECATED. class ModuleRtpRtcpImpl : public RtpRtcp, public RTCPReceiver::ModuleRtpRtcp { public: explicit ModuleRtpRtcpImpl( const RtpRtcpInterface::Configuration& configuration); ~ModuleRtpRtcpImpl() override; // Returns the number of milliseconds until the module want a worker thread to // call Process. int64_t TimeUntilNextProcess() override; // Process any pending tasks such as timeouts. void Process() override; // Receiver part. // Called when we receive an RTCP packet. void IncomingRtcpPacket(const uint8_t* incoming_packet, size_t incoming_packet_length) override; void SetRemoteSSRC(uint32_t 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; // Register RTP header extension. void RegisterRtpHeaderExtension(absl::string_view uri, int id) override; int32_t DeregisterSendRtpHeaderExtension(RTPExtensionType type) 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; uint32_t SSRC() const override { return rtcp_sender_.SSRC(); } void SetRid(const std::string& rid) override; void SetMid(const std::string& mid) override; void SetCsrcs(const std::vector& csrcs) override; RTCPSender::FeedbackState GetFeedbackState(); void SetRtxSendStatus(int mode) override; int RtxSendStatus() const override; absl::optional RtxSsrc() const override; void SetRtxSendPayloadType(int payload_type, int associated_payload_type) override; absl::optional 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 TrySendPacket(RtpPacketToSend* packet, const PacedPacketInfo& pacing_info) override; void SetFecProtectionParams(const FecProtectionParams& delta_params, const FecProtectionParams& key_params) override; std::vector> FetchFecPackets() override; void OnPacketsAcknowledged( rtc::ArrayView sequence_numbers) override; std::vector> GeneratePadding( size_t target_size_bytes) override; std::vector GetSentRtpPacketInfos( rtc::ArrayView sequence_numbers) const override; size_t ExpectedPerPacketOverhead() const 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(const char* c_name) override; // Get remote NTP. int32_t RemoteNTP(uint32_t* received_ntp_secs, uint32_t* received_ntp_frac, uint32_t* rtcp_arrival_time_secs, uint32_t* rtcp_arrival_time_frac, uint32_t* rtcp_timestamp) const override; // Get RoundTripTime. int32_t RTT(uint32_t remote_ssrc, int64_t* rtt, int64_t* avg_rtt, int64_t* min_rtt, int64_t* max_rtt) const override; int64_t ExpectedRetransmissionTimeMs() const override; // Force a send of an RTCP packet. // Normal SR and RR are triggered via the process function. 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::RTCPReportBlock::source_ssrc(), // which is the SSRC of the corresponding outbound RTP stream, is unique. std::vector GetLatestReportBlockData() const override; absl::optional GetSenderReportStats() const override; // (REMB) Receiver Estimated Max Bitrate. void SetRemb(int64_t bitrate_bps, std::vector ssrcs) override; void UnsetRemb() override; void SetTmmbn(std::vector 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& 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> 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& nack_sequence_numbers) override; void OnReceivedRtcpReportBlocks( const ReportBlockList& report_blocks) override; void OnRequestSendReport() override; void SetVideoBitrateAllocation( const VideoBitrateAllocation& bitrate) override; RTPSender* RtpSender() override; const RTPSender* RtpSender() const override; protected: bool UpdateRTCPReceiveInformationTimers(); RTPSender* rtp_sender() { return rtp_sender_ ? &rtp_sender_->packet_generator : nullptr; } const RTPSender* rtp_sender() const { return rtp_sender_ ? &rtp_sender_->packet_generator : nullptr; } RTCPSender* rtcp_sender() { return &rtcp_sender_; } const RTCPSender* rtcp_sender() const { return &rtcp_sender_; } RTCPReceiver* rtcp_receiver() { return &rtcp_receiver_; } const RTCPReceiver* rtcp_receiver() const { return &rtcp_receiver_; } void SetMediaHasBeenSent(bool media_has_been_sent) { rtp_sender_->packet_sender.SetMediaHasBeenSent(media_has_been_sent); } Clock* clock() const { return clock_; } private: FRIEND_TEST_ALL_PREFIXES(RtpRtcpImplTest, Rtt); FRIEND_TEST_ALL_PREFIXES(RtpRtcpImplTest, RttForReceiverOnly); struct RtpSenderContext { explicit RtpSenderContext(const RtpRtcpInterface::Configuration& config); // Storage of packets, for retransmissions and padding, if applicable. RtpPacketHistory packet_history; // Handles final time timestamping/stats/etc and handover to Transport. DEPRECATED_RtpSenderEgress packet_sender; // If no paced sender configured, this class will be used to pass packets // from |packet_generator_| to |packet_sender_|. DEPRECATED_RtpSenderEgress::NonPacedPacketSender non_paced_sender; // Handles creation of RTP packets to be sent. RTPSender packet_generator; }; void set_rtt_ms(int64_t rtt_ms); int64_t rtt_ms() const; bool TimeToSendFullNackList(int64_t now) const; // Returns true if the module is configured to store packets. bool StorePackets() const; // Returns current Receiver Reference Time Report (RTTR) status. bool RtcpXrRrtrStatus() const; std::unique_ptr rtp_sender_; RTCPSender rtcp_sender_; RTCPReceiver rtcp_receiver_; Clock* const clock_; int64_t last_bitrate_process_time_; int64_t last_rtt_process_time_; int64_t next_process_time_; uint16_t packet_overhead_; // Send side int64_t nack_last_time_sent_full_ms_; uint16_t nack_last_seq_number_sent_; RemoteBitrateEstimator* const remote_bitrate_; RtcpRttStats* const rtt_stats_; // The processed RTT from RtcpRttStats. mutable Mutex mutex_rtt_; int64_t rtt_ms_ RTC_GUARDED_BY(mutex_rtt_); }; } // namespace webrtc #endif // MODULES_RTP_RTCP_SOURCE_RTP_RTCP_IMPL_H_