| /* |
| * 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 WEBRTC_MODULES_RTP_RTCP_SOURCE_RTCP_SENDER_H_ |
| #define WEBRTC_MODULES_RTP_RTCP_SOURCE_RTCP_SENDER_H_ |
| |
| #include <map> |
| #include <set> |
| #include <sstream> |
| #include <string> |
| |
| #include "webrtc/base/scoped_ptr.h" |
| #include "webrtc/base/thread_annotations.h" |
| #include "webrtc/modules/remote_bitrate_estimator/include/bwe_defines.h" |
| #include "webrtc/modules/remote_bitrate_estimator/include/remote_bitrate_estimator.h" |
| #include "webrtc/modules/rtp_rtcp/interface/receive_statistics.h" |
| #include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h" |
| #include "webrtc/modules/rtp_rtcp/source/rtcp_packet.h" |
| #include "webrtc/modules/rtp_rtcp/source/rtcp_utility.h" |
| #include "webrtc/modules/rtp_rtcp/source/rtp_utility.h" |
| #include "webrtc/modules/rtp_rtcp/source/tmmbr_help.h" |
| #include "webrtc/typedefs.h" |
| |
| namespace webrtc { |
| |
| class ModuleRtpRtcpImpl; |
| class RTCPReceiver; |
| |
| class NACKStringBuilder { |
| public: |
| NACKStringBuilder(); |
| ~NACKStringBuilder(); |
| |
| void PushNACK(uint16_t nack); |
| std::string GetResult(); |
| |
| private: |
| std::ostringstream stream_; |
| int count_; |
| uint16_t prevNack_; |
| bool consecutive_; |
| }; |
| |
| class RTCPSender { |
| public: |
| struct FeedbackState { |
| FeedbackState(); |
| |
| uint8_t send_payload_type; |
| uint32_t frequency_hz; |
| uint32_t packets_sent; |
| size_t media_bytes_sent; |
| uint32_t send_bitrate; |
| |
| uint32_t last_rr_ntp_secs; |
| uint32_t last_rr_ntp_frac; |
| uint32_t remote_sr; |
| |
| bool has_last_xr_rr; |
| RtcpReceiveTimeInfo last_xr_rr; |
| |
| // Used when generating TMMBR. |
| ModuleRtpRtcpImpl* module; |
| }; |
| |
| RTCPSender(int32_t id, |
| bool audio, |
| Clock* clock, |
| ReceiveStatistics* receive_statistics, |
| RtcpPacketTypeCounterObserver* packet_type_counter_observer); |
| virtual ~RTCPSender(); |
| |
| int32_t RegisterSendTransport(Transport* outgoingTransport); |
| |
| RTCPMethod Status() const; |
| void SetRTCPStatus(RTCPMethod method); |
| |
| bool Sending() const; |
| int32_t SetSendingStatus(const FeedbackState& feedback_state, |
| bool enabled); // combine the functions |
| |
| int32_t SetNackStatus(bool enable); |
| |
| void SetStartTimestamp(uint32_t start_timestamp); |
| |
| void SetLastRtpTime(uint32_t rtp_timestamp, int64_t capture_time_ms); |
| |
| void SetSSRC(uint32_t ssrc); |
| |
| void SetRemoteSSRC(uint32_t ssrc); |
| |
| int32_t SetCNAME(const char* cName); |
| |
| int32_t AddMixedCNAME(uint32_t SSRC, const char cName[RTCP_CNAME_SIZE]); |
| |
| int32_t RemoveMixedCNAME(uint32_t SSRC); |
| |
| int64_t SendTimeOfSendReport(uint32_t sendReport); |
| |
| bool SendTimeOfXrRrReport(uint32_t mid_ntp, int64_t* time_ms) const; |
| |
| bool TimeToSendRTCPReport(bool sendKeyframeBeforeRTP = false) const; |
| |
| int32_t SendRTCP(const FeedbackState& feedback_state, |
| RTCPPacketType packetType, |
| int32_t nackSize = 0, |
| const uint16_t* nackList = 0, |
| bool repeat = false, |
| uint64_t pictureID = 0); |
| |
| int32_t SendCompoundRTCP(const FeedbackState& feedback_state, |
| const std::set<RTCPPacketType>& packetTypes, |
| int32_t nackSize = 0, |
| const uint16_t* nackList = 0, |
| bool repeat = false, |
| uint64_t pictureID = 0); |
| |
| bool REMB() const; |
| |
| void SetREMBStatus(bool enable); |
| |
| void SetREMBData(uint32_t bitrate, const std::vector<uint32_t>& ssrcs); |
| |
| bool TMMBR() const; |
| |
| void SetTMMBRStatus(bool enable); |
| |
| int32_t SetTMMBN(const TMMBRSet* boundingSet, uint32_t maxBitrateKbit); |
| |
| // Extended jitter report |
| bool IJ() const; |
| |
| void SetIJStatus(bool enable); |
| |
| int32_t SetApplicationSpecificData(uint8_t subType, |
| uint32_t name, |
| const uint8_t* data, |
| uint16_t length); |
| int32_t SetRTCPVoIPMetrics(const RTCPVoIPMetric* VoIPMetric); |
| |
| void SendRtcpXrReceiverReferenceTime(bool enable); |
| |
| bool RtcpXrReceiverReferenceTime() const; |
| |
| void SetCsrcs(const std::vector<uint32_t>& csrcs); |
| |
| void SetTargetBitrate(unsigned int target_bitrate); |
| |
| private: |
| struct RtcpContext; |
| |
| // The BuildResult indicates the outcome of a call to a builder method, |
| // constructing a part of an RTCP packet: |
| // |
| // kError |
| // Building RTCP packet failed, propagate error out to caller. |
| // kAbort |
| // The (partial) block being build should not be included. Reset current |
| // buffer position to the state before the method call and proceed to the |
| // next packet type. |
| // kTruncated |
| // There is not enough room in the buffer to fit the data being constructed. |
| // (IP packet is full). Proceed to the next packet type, and call this |
| // method again when a new buffer has been allocated. |
| // TODO(sprang): Actually allocate multiple packets if needed. |
| // kSuccess |
| // Data has been successfully placed in the buffer. |
| |
| enum class BuildResult { kError, kAborted, kTruncated, kSuccess }; |
| |
| int32_t SendToNetwork(const uint8_t* dataBuffer, size_t length); |
| |
| int32_t AddReportBlock(const RTCPReportBlock& report_block) |
| EXCLUSIVE_LOCKS_REQUIRED(critical_section_rtcp_sender_); |
| |
| bool PrepareReport(const FeedbackState& feedback_state, |
| uint32_t ssrc, |
| StreamStatistician* statistician, |
| RTCPReportBlock* report_block); |
| |
| int PrepareRTCP(const FeedbackState& feedback_state, |
| const std::set<RTCPPacketType>& packetTypes, |
| int32_t nackSize, |
| const uint16_t* nackList, |
| bool repeat, |
| uint64_t pictureID, |
| uint8_t* rtcp_buffer, |
| int buffer_size); |
| |
| BuildResult BuildSR(RtcpContext* context) |
| EXCLUSIVE_LOCKS_REQUIRED(critical_section_rtcp_sender_); |
| BuildResult BuildRR(RtcpContext* context) |
| EXCLUSIVE_LOCKS_REQUIRED(critical_section_rtcp_sender_); |
| BuildResult BuildExtendedJitterReport(RtcpContext* context) |
| EXCLUSIVE_LOCKS_REQUIRED(critical_section_rtcp_sender_); |
| BuildResult BuildSDEC(RtcpContext* context) |
| EXCLUSIVE_LOCKS_REQUIRED(critical_section_rtcp_sender_); |
| BuildResult BuildPLI(RtcpContext* context) |
| EXCLUSIVE_LOCKS_REQUIRED(critical_section_rtcp_sender_); |
| BuildResult BuildREMB(RtcpContext* context) |
| EXCLUSIVE_LOCKS_REQUIRED(critical_section_rtcp_sender_); |
| BuildResult BuildTMMBR(RtcpContext* context) |
| EXCLUSIVE_LOCKS_REQUIRED(critical_section_rtcp_sender_); |
| BuildResult BuildTMMBN(RtcpContext* context) |
| EXCLUSIVE_LOCKS_REQUIRED(critical_section_rtcp_sender_); |
| BuildResult BuildAPP(RtcpContext* context) |
| EXCLUSIVE_LOCKS_REQUIRED(critical_section_rtcp_sender_); |
| BuildResult BuildVoIPMetric(RtcpContext* context) |
| EXCLUSIVE_LOCKS_REQUIRED(critical_section_rtcp_sender_); |
| BuildResult BuildBYE(RtcpContext* context) |
| EXCLUSIVE_LOCKS_REQUIRED(critical_section_rtcp_sender_); |
| BuildResult BuildFIR(RtcpContext* context) |
| EXCLUSIVE_LOCKS_REQUIRED(critical_section_rtcp_sender_); |
| BuildResult BuildSLI(RtcpContext* context) |
| EXCLUSIVE_LOCKS_REQUIRED(critical_section_rtcp_sender_); |
| BuildResult BuildRPSI(RtcpContext* context) |
| EXCLUSIVE_LOCKS_REQUIRED(critical_section_rtcp_sender_); |
| BuildResult BuildNACK(RtcpContext* context) |
| EXCLUSIVE_LOCKS_REQUIRED(critical_section_rtcp_sender_); |
| BuildResult BuildReceiverReferenceTime(RtcpContext* context) |
| EXCLUSIVE_LOCKS_REQUIRED(critical_section_rtcp_sender_); |
| BuildResult BuildDlrr(RtcpContext* context) |
| EXCLUSIVE_LOCKS_REQUIRED(critical_section_rtcp_sender_); |
| |
| private: |
| const int32_t id_; |
| const bool audio_; |
| Clock* const clock_; |
| RTCPMethod method_ GUARDED_BY(critical_section_rtcp_sender_); |
| |
| rtc::scoped_ptr<CriticalSectionWrapper> critical_section_transport_; |
| Transport* cbTransport_ GUARDED_BY(critical_section_transport_); |
| |
| rtc::scoped_ptr<CriticalSectionWrapper> critical_section_rtcp_sender_; |
| bool using_nack_ GUARDED_BY(critical_section_rtcp_sender_); |
| bool sending_ GUARDED_BY(critical_section_rtcp_sender_); |
| bool remb_enabled_ GUARDED_BY(critical_section_rtcp_sender_); |
| bool extended_jitter_report_enabled_ GUARDED_BY(critical_section_rtcp_sender_); |
| |
| int64_t next_time_to_send_rtcp_ GUARDED_BY(critical_section_rtcp_sender_); |
| |
| uint32_t start_timestamp_ GUARDED_BY(critical_section_rtcp_sender_); |
| uint32_t last_rtp_timestamp_ GUARDED_BY(critical_section_rtcp_sender_); |
| int64_t last_frame_capture_time_ms_ GUARDED_BY(critical_section_rtcp_sender_); |
| uint32_t ssrc_ GUARDED_BY(critical_section_rtcp_sender_); |
| // SSRC that we receive on our RTP channel |
| uint32_t remote_ssrc_ GUARDED_BY(critical_section_rtcp_sender_); |
| char cname_[RTCP_CNAME_SIZE] GUARDED_BY(critical_section_rtcp_sender_); |
| |
| ReceiveStatistics* receive_statistics_ |
| GUARDED_BY(critical_section_rtcp_sender_); |
| std::map<uint32_t, rtcp::ReportBlock> report_blocks_ |
| GUARDED_BY(critical_section_rtcp_sender_); |
| // TODO(sprang): Can we avoid pointers here? |
| std::map<uint32_t, RTCPUtility::RTCPCnameInformation*> csrc_cnames_ |
| GUARDED_BY(critical_section_rtcp_sender_); |
| |
| // Sent |
| uint32_t last_send_report_[RTCP_NUMBER_OF_SR] GUARDED_BY( |
| critical_section_rtcp_sender_); // allow packet loss and RTT above 1 sec |
| int64_t last_rtcp_time_[RTCP_NUMBER_OF_SR] GUARDED_BY( |
| critical_section_rtcp_sender_); |
| |
| // Sent XR receiver reference time report. |
| // <mid ntp (mid 32 bits of the 64 bits NTP timestamp), send time in ms>. |
| std::map<uint32_t, int64_t> last_xr_rr_ |
| GUARDED_BY(critical_section_rtcp_sender_); |
| |
| // send CSRCs |
| std::vector<uint32_t> csrcs_ GUARDED_BY(critical_section_rtcp_sender_); |
| |
| // Full intra request |
| uint8_t sequence_number_fir_ GUARDED_BY(critical_section_rtcp_sender_); |
| |
| // REMB |
| uint32_t remb_bitrate_ GUARDED_BY(critical_section_rtcp_sender_); |
| std::vector<uint32_t> remb_ssrcs_ GUARDED_BY(critical_section_rtcp_sender_); |
| |
| TMMBRHelp tmmbr_help_ GUARDED_BY(critical_section_rtcp_sender_); |
| uint32_t tmmbr_send_ GUARDED_BY(critical_section_rtcp_sender_); |
| uint32_t packet_oh_send_ GUARDED_BY(critical_section_rtcp_sender_); |
| |
| // APP |
| uint8_t app_sub_type_ GUARDED_BY(critical_section_rtcp_sender_); |
| uint32_t app_name_ GUARDED_BY(critical_section_rtcp_sender_); |
| rtc::scoped_ptr<uint8_t[]> app_data_ GUARDED_BY(critical_section_rtcp_sender_); |
| uint16_t app_length_ GUARDED_BY(critical_section_rtcp_sender_); |
| |
| // True if sending of XR Receiver reference time report is enabled. |
| bool xr_send_receiver_reference_time_enabled_ |
| GUARDED_BY(critical_section_rtcp_sender_); |
| |
| // XR VoIP metric |
| RTCPVoIPMetric xr_voip_metric_ GUARDED_BY(critical_section_rtcp_sender_); |
| |
| RtcpPacketTypeCounterObserver* const packet_type_counter_observer_; |
| RtcpPacketTypeCounter packet_type_counter_ |
| GUARDED_BY(critical_section_rtcp_sender_); |
| |
| RTCPUtility::NackStats nack_stats_ GUARDED_BY(critical_section_rtcp_sender_); |
| |
| void SetFlag(RTCPPacketType type, bool is_volatile) |
| EXCLUSIVE_LOCKS_REQUIRED(critical_section_rtcp_sender_); |
| void SetFlags(const std::set<RTCPPacketType>& types, bool is_volatile) |
| EXCLUSIVE_LOCKS_REQUIRED(critical_section_rtcp_sender_); |
| bool IsFlagPresent(RTCPPacketType type) const |
| EXCLUSIVE_LOCKS_REQUIRED(critical_section_rtcp_sender_); |
| bool ConsumeFlag(RTCPPacketType type, bool forced = false) |
| EXCLUSIVE_LOCKS_REQUIRED(critical_section_rtcp_sender_); |
| bool AllVolatileFlagsConsumed() const |
| EXCLUSIVE_LOCKS_REQUIRED(critical_section_rtcp_sender_); |
| struct ReportFlag { |
| ReportFlag(RTCPPacketType type, bool is_volatile) |
| : type(type), is_volatile(is_volatile) {} |
| bool operator<(const ReportFlag& flag) const { return type < flag.type; } |
| bool operator==(const ReportFlag& flag) const { return type == flag.type; } |
| const RTCPPacketType type; |
| const bool is_volatile; |
| }; |
| |
| std::set<ReportFlag> report_flags_ GUARDED_BY(critical_section_rtcp_sender_); |
| |
| typedef BuildResult (RTCPSender::*Builder)(RtcpContext*); |
| std::map<RTCPPacketType, Builder> builders_; |
| |
| class PacketBuiltCallback; |
| }; |
| } // namespace webrtc |
| |
| #endif // WEBRTC_MODULES_RTP_RTCP_SOURCE_RTCP_SENDER_H_ |