| /* |
| * 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 <stddef.h> |
| #include <stdint.h> |
| |
| #include <memory> |
| #include <set> |
| #include <string> |
| #include <vector> |
| |
| #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; |
| void SetLocalSsrc(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; |
| |
| void SetNonSenderRttMeasurement(bool enabled) 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<uint32_t>& csrcs) 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 TrySendPacket(RtpPacketToSend* packet, |
| const PacedPacketInfo& pacing_info) override; |
| |
| void SetFecProtectionParams(const FecProtectionParams& delta_params, |
| const FecProtectionParams& key_params) override; |
| |
| std::vector<std::unique_ptr<RtpPacketToSend>> FetchFecPackets() 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; |
| |
| // 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<ReportBlockData> GetLatestReportBlockData() const override; |
| absl::optional<SenderReportStats> GetSenderReportStats() 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( |
| 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<RtpSenderContext> 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_ |