| /* |
| * 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_INCLUDE_RTP_RTCP_H_ |
| #define WEBRTC_MODULES_RTP_RTCP_INCLUDE_RTP_RTCP_H_ |
| |
| #include <set> |
| #include <string> |
| #include <utility> |
| #include <vector> |
| |
| #include "webrtc/base/constructormagic.h" |
| #include "webrtc/modules/include/module.h" |
| #include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h" |
| #include "webrtc/modules/video_coding/include/video_coding_defines.h" |
| |
| namespace webrtc { |
| |
| // Forward declarations. |
| class RateLimiter; |
| class ReceiveStatistics; |
| class RemoteBitrateEstimator; |
| class RtcEventLog; |
| class RtpReceiver; |
| class Transport; |
| |
| RTPExtensionType StringToRtpExtensionType(const std::string& extension); |
| |
| namespace rtcp { |
| class TransportFeedback; |
| } |
| |
| class RtpRtcp : public Module { |
| public: |
| struct Configuration { |
| Configuration(); |
| |
| // True for a audio version of the RTP/RTCP module object false will create |
| // a video version. |
| bool audio = false; |
| bool receiver_only = false; |
| |
| // The clock to use to read time. If nullptr then system clock will be used. |
| Clock* clock = nullptr; |
| |
| ReceiveStatistics* receive_statistics; |
| |
| // Transport object that will be called when packets are ready to be sent |
| // out on the network. |
| Transport* outgoing_transport = nullptr; |
| |
| // Called when the receiver request a intra frame. |
| RtcpIntraFrameObserver* intra_frame_callback = nullptr; |
| |
| // Called when we receive a changed estimate from the receiver of out |
| // stream. |
| RtcpBandwidthObserver* bandwidth_callback = nullptr; |
| |
| TransportFeedbackObserver* transport_feedback_callback = nullptr; |
| RtcpRttStats* rtt_stats = nullptr; |
| RtcpPacketTypeCounterObserver* rtcp_packet_type_counter_observer = nullptr; |
| |
| // Estimates the bandwidth available for a set of streams from the same |
| // client. |
| RemoteBitrateEstimator* remote_bitrate_estimator = nullptr; |
| |
| // Spread any bursts of packets into smaller bursts to minimize packet loss. |
| RtpPacketSender* paced_sender = nullptr; |
| |
| TransportSequenceNumberAllocator* transport_sequence_number_allocator = |
| nullptr; |
| BitrateStatisticsObserver* send_bitrate_observer = nullptr; |
| FrameCountObserver* send_frame_count_observer = nullptr; |
| SendSideDelayObserver* send_side_delay_observer = nullptr; |
| RtcEventLog* event_log = nullptr; |
| SendPacketObserver* send_packet_observer = nullptr; |
| RateLimiter* retransmission_rate_limiter = nullptr; |
| |
| private: |
| RTC_DISALLOW_COPY_AND_ASSIGN(Configuration); |
| }; |
| |
| // Create a RTP/RTCP module object using the system clock. |
| // |configuration| - Configuration of the RTP/RTCP module. |
| static RtpRtcp* CreateRtpRtcp(const RtpRtcp::Configuration& configuration); |
| |
| // ************************************************************************** |
| // Receiver functions |
| // ************************************************************************** |
| |
| virtual int32_t IncomingRtcpPacket(const uint8_t* incoming_packet, |
| size_t incoming_packet_length) = 0; |
| |
| virtual void SetRemoteSSRC(uint32_t ssrc) = 0; |
| |
| // ************************************************************************** |
| // Sender |
| // ************************************************************************** |
| |
| // Sets MTU. |
| // |size| - Max transfer unit in bytes, default is 1500. |
| // Returns -1 on failure else 0. |
| virtual int32_t SetMaxTransferUnit(uint16_t size) = 0; |
| |
| // Sets transtport overhead. Default is IPv4 and UDP with no encryption. |
| // |tcp| - true for TCP false UDP. |
| // |ipv6| - true for IP version 6 false for version 4. |
| // |authentication_overhead| - number of bytes to leave for an authentication |
| // header. |
| // Returns -1 on failure else 0 |
| virtual int32_t SetTransportOverhead(bool tcp, |
| bool ipv6, |
| uint8_t authentication_overhead = 0) = 0; |
| |
| // Returns max payload length, which is a combination of the configuration |
| // MaxTransferUnit and TransportOverhead. |
| // Does not account for RTP headers and FEC/ULP/RED overhead (when FEC is |
| // enabled). |
| virtual uint16_t MaxPayloadLength() const = 0; |
| |
| // Returns max data payload length, which is a combination of the |
| // configuration MaxTransferUnit, headers and TransportOverhead. |
| // Takes into account RTP headers and FEC/ULP/RED overhead (when FEC is |
| // enabled). |
| virtual uint16_t MaxDataPayloadLength() const = 0; |
| |
| // Sets codec name and payload type. Returns -1 on failure else 0. |
| virtual int32_t RegisterSendPayload(const CodecInst& voice_codec) = 0; |
| |
| // Sets codec name and payload type. Return -1 on failure else 0. |
| virtual int32_t RegisterSendPayload(const VideoCodec& video_codec) = 0; |
| |
| virtual void RegisterVideoSendPayload(int payload_type, |
| const char* payload_name) = 0; |
| |
| // Unregisters a send payload. |
| // |payload_type| - payload type of codec |
| // Returns -1 on failure else 0. |
| virtual int32_t DeRegisterSendPayload(int8_t payload_type) = 0; |
| |
| // (De)registers RTP header extension type and id. |
| // Returns -1 on failure else 0. |
| virtual int32_t RegisterSendRtpHeaderExtension(RTPExtensionType type, |
| uint8_t id) = 0; |
| |
| virtual int32_t DeregisterSendRtpHeaderExtension(RTPExtensionType type) = 0; |
| |
| // Returns start timestamp. |
| virtual uint32_t StartTimestamp() const = 0; |
| |
| // Sets start timestamp. Start timestamp is set to a random value if this |
| // function is never called. |
| virtual void SetStartTimestamp(uint32_t timestamp) = 0; |
| |
| // Returns SequenceNumber. |
| virtual uint16_t SequenceNumber() const = 0; |
| |
| // Sets SequenceNumber, default is a random number. |
| virtual void SetSequenceNumber(uint16_t seq) = 0; |
| |
| virtual void SetRtpState(const RtpState& rtp_state) = 0; |
| virtual void SetRtxState(const RtpState& rtp_state) = 0; |
| virtual RtpState GetRtpState() const = 0; |
| virtual RtpState GetRtxState() const = 0; |
| |
| // Returns SSRC. |
| virtual uint32_t SSRC() const = 0; |
| |
| // Sets SSRC, default is a random number. |
| virtual void SetSSRC(uint32_t ssrc) = 0; |
| |
| // Sets CSRC. |
| // |csrcs| - vector of CSRCs |
| virtual void SetCsrcs(const std::vector<uint32_t>& csrcs) = 0; |
| |
| // Turns on/off sending RTX (RFC 4588). The modes can be set as a combination |
| // of values of the enumerator RtxMode. |
| virtual void SetRtxSendStatus(int modes) = 0; |
| |
| // Returns status of sending RTX (RFC 4588). The returned value can be |
| // a combination of values of the enumerator RtxMode. |
| virtual int RtxSendStatus() const = 0; |
| |
| // Sets the SSRC to use when sending RTX packets. This doesn't enable RTX, |
| // only the SSRC is set. |
| virtual void SetRtxSsrc(uint32_t ssrc) = 0; |
| |
| // Sets the payload type to use when sending RTX packets. Note that this |
| // doesn't enable RTX, only the payload type is set. |
| virtual void SetRtxSendPayloadType(int payload_type, |
| int associated_payload_type) = 0; |
| |
| // Sets sending status. Sends kRtcpByeCode when going from true to false. |
| // Returns -1 on failure else 0. |
| virtual int32_t SetSendingStatus(bool sending) = 0; |
| |
| // Returns current sending status. |
| virtual bool Sending() const = 0; |
| |
| // Starts/Stops media packets. On by default. |
| virtual void SetSendingMediaStatus(bool sending) = 0; |
| |
| // Returns current media sending status. |
| virtual bool SendingMedia() const = 0; |
| |
| // Returns current bitrate in Kbit/s. |
| virtual void BitrateSent(uint32_t* total_rate, |
| uint32_t* video_rate, |
| uint32_t* fec_rate, |
| uint32_t* nack_rate) const = 0; |
| |
| // Used by the codec module to deliver a video or audio frame for |
| // packetization. |
| // |frame_type| - type of frame to send |
| // |payload_type| - payload type of frame to send |
| // |timestamp| - timestamp of frame to send |
| // |payload_data| - payload buffer of frame to send |
| // |payload_size| - size of payload buffer to send |
| // |fragmentation| - fragmentation offset data for fragmented frames such |
| // as layers or RED |
| // |transport_frame_id_out| - set to RTP timestamp. |
| // Returns true on success. |
| |
| virtual bool SendOutgoingData(FrameType frame_type, |
| int8_t payload_type, |
| uint32_t timestamp, |
| int64_t capture_time_ms, |
| const uint8_t* payload_data, |
| size_t payload_size, |
| const RTPFragmentationHeader* fragmentation, |
| const RTPVideoHeader* rtp_video_header, |
| uint32_t* transport_frame_id_out) = 0; |
| |
| // Deprecated version of the method above. |
| int32_t SendOutgoingData( |
| FrameType frame_type, |
| int8_t payload_type, |
| uint32_t timestamp, |
| int64_t capture_time_ms, |
| const uint8_t* payload_data, |
| size_t payload_size, |
| const RTPFragmentationHeader* fragmentation = nullptr, |
| const RTPVideoHeader* rtp_video_header = nullptr) { |
| return SendOutgoingData(frame_type, payload_type, timestamp, |
| capture_time_ms, payload_data, payload_size, |
| fragmentation, rtp_video_header, |
| /*frame_id_out=*/nullptr) |
| ? 0 |
| : -1; |
| } |
| |
| virtual bool TimeToSendPacket(uint32_t ssrc, |
| uint16_t sequence_number, |
| int64_t capture_time_ms, |
| bool retransmission, |
| int probe_cluster_id) = 0; |
| |
| virtual size_t TimeToSendPadding(size_t bytes, int probe_cluster_id) = 0; |
| |
| // Called on generation of new statistics after an RTP send. |
| virtual void RegisterSendChannelRtpStatisticsCallback( |
| StreamDataCountersCallback* callback) = 0; |
| virtual StreamDataCountersCallback* GetSendChannelRtpStatisticsCallback() |
| const = 0; |
| |
| // ************************************************************************** |
| // RTCP |
| // ************************************************************************** |
| |
| // Returns RTCP status. |
| virtual RtcpMode RTCP() const = 0; |
| |
| // Sets RTCP status i.e on(compound or non-compound)/off. |
| // |method| - RTCP method to use. |
| virtual void SetRTCPStatus(RtcpMode method) = 0; |
| |
| // Sets RTCP CName (i.e unique identifier). |
| // Returns -1 on failure else 0. |
| virtual int32_t SetCNAME(const char* cname) = 0; |
| |
| // Returns remote CName. |
| // Returns -1 on failure else 0. |
| virtual int32_t RemoteCNAME(uint32_t remote_ssrc, |
| char cname[RTCP_CNAME_SIZE]) const = 0; |
| |
| // Returns remote NTP. |
| // Returns -1 on failure else 0. |
| virtual 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 = 0; |
| |
| // Returns -1 on failure else 0. |
| virtual int32_t AddMixedCNAME(uint32_t ssrc, const char* cname) = 0; |
| |
| // Returns -1 on failure else 0. |
| virtual int32_t RemoveMixedCNAME(uint32_t ssrc) = 0; |
| |
| // Returns current RTT (round-trip time) estimate. |
| // Returns -1 on failure else 0. |
| virtual int32_t RTT(uint32_t remote_ssrc, |
| int64_t* rtt, |
| int64_t* avg_rtt, |
| int64_t* min_rtt, |
| int64_t* max_rtt) const = 0; |
| |
| // Forces a send of a RTCP packet. Periodic SR and RR are triggered via the |
| // process function. |
| // Returns -1 on failure else 0. |
| virtual int32_t SendRTCP(RTCPPacketType rtcp_packet_type) = 0; |
| |
| // Forces a send of a RTCP packet with more than one packet type. |
| // periodic SR and RR are triggered via the process function |
| // Returns -1 on failure else 0. |
| virtual int32_t SendCompoundRTCP( |
| const std::set<RTCPPacketType>& rtcp_packet_types) = 0; |
| |
| // Notifies the sender about good state of the RTP receiver. |
| virtual int32_t SendRTCPReferencePictureSelection(uint64_t picture_id) = 0; |
| |
| // Send a RTCP Slice Loss Indication (SLI). |
| // |picture_id| - 6 least significant bits of picture_id. |
| virtual int32_t SendRTCPSliceLossIndication(uint8_t picture_id) = 0; |
| |
| // Returns statistics of the amount of data sent. |
| // Returns -1 on failure else 0. |
| virtual int32_t DataCountersRTP(size_t* bytes_sent, |
| uint32_t* packets_sent) const = 0; |
| |
| // Returns send statistics for the RTP and RTX stream. |
| virtual void GetSendStreamDataCounters( |
| StreamDataCounters* rtp_counters, |
| StreamDataCounters* rtx_counters) const = 0; |
| |
| // Returns packet loss statistics for the RTP stream. |
| virtual void GetRtpPacketLossStats( |
| bool outgoing, |
| uint32_t ssrc, |
| struct RtpPacketLossStats* loss_stats) const = 0; |
| |
| // Returns received RTCP sender info. |
| // Returns -1 on failure else 0. |
| virtual int32_t RemoteRTCPStat(RTCPSenderInfo* sender_info) = 0; |
| |
| // Returns received RTCP report block. |
| // Returns -1 on failure else 0. |
| virtual int32_t RemoteRTCPStat( |
| std::vector<RTCPReportBlock>* receive_blocks) const = 0; |
| |
| // (APP) Sets application specific data. |
| // Returns -1 on failure else 0. |
| virtual int32_t SetRTCPApplicationSpecificData(uint8_t sub_type, |
| uint32_t name, |
| const uint8_t* data, |
| uint16_t length) = 0; |
| // (XR) Sets VOIP metric. |
| // Returns -1 on failure else 0. |
| virtual int32_t SetRTCPVoIPMetrics(const RTCPVoIPMetric* VoIPMetric) = 0; |
| |
| // (XR) Sets Receiver Reference Time Report (RTTR) status. |
| virtual void SetRtcpXrRrtrStatus(bool enable) = 0; |
| |
| // Returns current Receiver Reference Time Report (RTTR) status. |
| virtual bool RtcpXrRrtrStatus() const = 0; |
| |
| // (REMB) Receiver Estimated Max Bitrate. |
| virtual bool REMB() const = 0; |
| |
| virtual void SetREMBStatus(bool enable) = 0; |
| |
| virtual void SetREMBData(uint32_t bitrate, |
| const std::vector<uint32_t>& ssrcs) = 0; |
| |
| // (TMMBR) Temporary Max Media Bit Rate |
| virtual bool TMMBR() const = 0; |
| |
| virtual void SetTMMBRStatus(bool enable) = 0; |
| |
| // (NACK) |
| |
| // TODO(holmer): Propagate this API to VideoEngine. |
| // Returns the currently configured selective retransmission settings. |
| virtual int SelectiveRetransmissions() const = 0; |
| |
| // TODO(holmer): Propagate this API to VideoEngine. |
| // Sets the selective retransmission settings, which will decide which |
| // packets will be retransmitted if NACKed. Settings are constructed by |
| // combining the constants in enum RetransmissionMode with bitwise OR. |
| // All packets are retransmitted if kRetransmitAllPackets is set, while no |
| // packets are retransmitted if kRetransmitOff is set. |
| // By default all packets except FEC packets are retransmitted. For VP8 |
| // with temporal scalability only base layer packets are retransmitted. |
| // Returns -1 on failure, otherwise 0. |
| virtual int SetSelectiveRetransmissions(uint8_t settings) = 0; |
| |
| // Sends a Negative acknowledgement packet. |
| // Returns -1 on failure else 0. |
| // TODO(philipel): Deprecate this and start using SendNack instead, mostly |
| // because we want a function that actually send NACK for the specified |
| // packets. |
| virtual int32_t SendNACK(const uint16_t* nack_list, uint16_t size) = 0; |
| |
| // Sends NACK for the packets specified. |
| // Note: This assumes the caller keeps track of timing and doesn't rely on |
| // the RTP module to do this. |
| virtual void SendNack(const std::vector<uint16_t>& sequence_numbers) = 0; |
| |
| // Store the sent packets, needed to answer to a Negative acknowledgment |
| // requests. |
| virtual void SetStorePacketsStatus(bool enable, uint16_t numberToStore) = 0; |
| |
| // Returns true if the module is configured to store packets. |
| virtual bool StorePackets() const = 0; |
| |
| // Called on receipt of RTCP report block from remote side. |
| virtual void RegisterRtcpStatisticsCallback( |
| RtcpStatisticsCallback* callback) = 0; |
| virtual RtcpStatisticsCallback* GetRtcpStatisticsCallback() = 0; |
| // BWE feedback packets. |
| virtual bool SendFeedbackPacket(const rtcp::TransportFeedback& packet) = 0; |
| |
| // ************************************************************************** |
| // Audio |
| // ************************************************************************** |
| |
| // Sets audio packet size, used to determine when it's time to send a DTMF |
| // packet in silence (CNG). |
| // Returns -1 on failure else 0. |
| virtual int32_t SetAudioPacketSize(uint16_t packet_size_samples) = 0; |
| |
| // Sends a TelephoneEvent tone using RFC 2833 (4733). |
| // Returns -1 on failure else 0. |
| virtual int32_t SendTelephoneEventOutband(uint8_t key, |
| uint16_t time_ms, |
| uint8_t level) = 0; |
| |
| // Store the audio level in dBov for header-extension-for-audio-level- |
| // indication. |
| // This API shall be called before transmision of an RTP packet to ensure |
| // that the |level| part of the extended RTP header is updated. |
| // return -1 on failure else 0. |
| virtual int32_t SetAudioLevel(uint8_t level_dbov) = 0; |
| |
| // ************************************************************************** |
| // Video |
| // ************************************************************************** |
| |
| // Turn on/off generic FEC. |
| virtual void SetGenericFECStatus(bool enable, |
| uint8_t payload_type_red, |
| uint8_t payload_type_fec) = 0; |
| |
| // Get generic FEC setting. |
| virtual void GenericFECStatus(bool* enable, |
| uint8_t* payload_type_red, |
| uint8_t* payload_type_fec) = 0; |
| |
| virtual int32_t SetFecParameters(const FecProtectionParams* delta_params, |
| const FecProtectionParams* key_params) = 0; |
| |
| // Set method for requestion a new key frame. |
| // Returns -1 on failure else 0. |
| virtual int32_t SetKeyFrameRequestMethod(KeyFrameRequestMethod method) = 0; |
| |
| // Sends a request for a keyframe. |
| // Returns -1 on failure else 0. |
| virtual int32_t RequestKeyFrame() = 0; |
| }; |
| |
| } // namespace webrtc |
| |
| #endif // WEBRTC_MODULES_RTP_RTCP_INCLUDE_RTP_RTCP_H_ |