| Index: webrtc/modules/rtp_rtcp/include/rtp_rtcp.h
|
| diff --git a/webrtc/modules/rtp_rtcp/include/rtp_rtcp.h b/webrtc/modules/rtp_rtcp/include/rtp_rtcp.h
|
| index bfd8e657432b2dc6197e9aab0d7f130a47d8f232..f0d23425bca70542525befb74952c8dee40ca971 100644
|
| --- a/webrtc/modules/rtp_rtcp/include/rtp_rtcp.h
|
| +++ b/webrtc/modules/rtp_rtcp/include/rtp_rtcp.h
|
| @@ -22,6 +22,7 @@
|
| #include "webrtc/modules/video_coding/include/video_coding_defines.h"
|
|
|
| namespace webrtc {
|
| +
|
| // Forward declarations.
|
| class RateLimiter;
|
| class ReceiveStatistics;
|
| @@ -41,613 +42,427 @@ class RtpRtcp : public Module {
|
| struct Configuration {
|
| Configuration();
|
|
|
| - /* id - Unique identifier of this RTP/RTCP module object
|
| - * audio - True for a audio version of the RTP/RTCP module
|
| - * object false will create a video version
|
| - * clock - The clock to use to read time. If NULL object
|
| - * will be using the system clock.
|
| - * incoming_data - Callback object that will receive the incoming
|
| - * data. May not be NULL; default callback will do
|
| - * nothing.
|
| - * incoming_messages - Callback object that will receive the incoming
|
| - * RTP messages. May not be NULL; default callback
|
| - * will do nothing.
|
| - * outgoing_transport - Transport object that will be called when packets
|
| - * are ready to be sent out on the network
|
| - * intra_frame_callback - Called when the receiver request a intra frame.
|
| - * bandwidth_callback - Called when we receive a changed estimate from
|
| - * the receiver of out stream.
|
| - * remote_bitrate_estimator - Estimates the bandwidth available for a set of
|
| - * streams from the same client.
|
| - * paced_sender - Spread any bursts of packets into smaller
|
| - * bursts to minimize packet loss.
|
| - */
|
| - bool audio;
|
| - bool receiver_only;
|
| - Clock* clock;
|
| + // 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* outgoing_transport;
|
| - RtcpIntraFrameObserver* intra_frame_callback;
|
| - RtcpBandwidthObserver* bandwidth_callback;
|
| - TransportFeedbackObserver* transport_feedback_callback;
|
| - RtcpRttStats* rtt_stats;
|
| - RtcpPacketTypeCounterObserver* rtcp_packet_type_counter_observer;
|
| - RemoteBitrateEstimator* remote_bitrate_estimator;
|
| - RtpPacketSender* paced_sender;
|
| - TransportSequenceNumberAllocator* transport_sequence_number_allocator;
|
| - BitrateStatisticsObserver* send_bitrate_observer;
|
| - FrameCountObserver* send_frame_count_observer;
|
| - SendSideDelayObserver* send_side_delay_observer;
|
| - RtcEventLog* event_log;
|
| - SendPacketObserver* send_packet_observer;
|
| - RateLimiter* retransmission_rate_limiter;
|
| +
|
| + // 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.
|
| - */
|
| + // 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
|
| - *
|
| - ***************************************************************************/
|
| -
|
| - /*
|
| - * set MTU
|
| - *
|
| - * size - Max transfer unit in bytes, default is 1500
|
| - *
|
| - * return -1 on failure else 0
|
| - */
|
| - virtual int32_t SetMaxTransferUnit(uint16_t size) = 0;
|
| -
|
| - /*
|
| - * set 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
|
| - * authenticationOverhead - number of bytes to leave for an
|
| - * authentication header
|
| - *
|
| - * return -1 on failure else 0
|
| - */
|
| - virtual int32_t SetTransportOverhead(
|
| - bool TCP,
|
| - bool IPV6,
|
| - uint8_t authenticationOverhead = 0) = 0;
|
| -
|
| - /*
|
| - * Get max payload length
|
| - *
|
| - * A combination of the configuration MaxTransferUnit and
|
| - * TransportOverhead.
|
| - * Does not account FEC/ULP/RED overhead if FEC is enabled.
|
| - * Does not account for RTP headers
|
| - */
|
| - virtual uint16_t MaxPayloadLength() const = 0;
|
| -
|
| - /*
|
| - * Get max data payload length
|
| - *
|
| - * A combination of the configuration MaxTransferUnit, headers and
|
| - * TransportOverhead.
|
| - * Takes into account FEC/ULP/RED overhead if FEC is enabled.
|
| - * Takes into account RTP headers
|
| - */
|
| - virtual uint16_t MaxDataPayloadLength() const = 0;
|
| -
|
| - /*
|
| - * set codec name and payload type
|
| - *
|
| - * return -1 on failure else 0
|
| - */
|
| - virtual int32_t RegisterSendPayload(
|
| - const CodecInst& voiceCodec) = 0;
|
| -
|
| - /*
|
| - * set codec name and payload type
|
| - *
|
| - * return -1 on failure else 0
|
| - */
|
| - virtual int32_t RegisterSendPayload(
|
| - const VideoCodec& videoCodec) = 0;
|
| -
|
| - virtual void RegisterVideoSendPayload(int payload_type,
|
| - const char* payload_name) = 0;
|
| -
|
| - /*
|
| - * Unregister a send payload
|
| - *
|
| - * payloadType - payload type of codec
|
| - *
|
| - * return -1 on failure else 0
|
| - */
|
| - virtual int32_t DeRegisterSendPayload(int8_t payloadType) = 0;
|
| -
|
| - /*
|
| - * (De)register RTP header extension type and id.
|
| - *
|
| - * return -1 on failure else 0
|
| - */
|
| - virtual int32_t RegisterSendRtpHeaderExtension(RTPExtensionType type,
|
| - uint8_t id) = 0;
|
| -
|
| - virtual int32_t DeregisterSendRtpHeaderExtension(RTPExtensionType type) = 0;
|
| -
|
| - /*
|
| - * get start timestamp
|
| - */
|
| - virtual uint32_t StartTimestamp() const = 0;
|
| -
|
| - /*
|
| - * configure start timestamp, default is a random number
|
| - *
|
| - * timestamp - start timestamp
|
| - */
|
| - virtual void SetStartTimestamp(uint32_t timestamp) = 0;
|
| -
|
| - /*
|
| - * Get SequenceNumber
|
| - */
|
| - virtual uint16_t SequenceNumber() const = 0;
|
| -
|
| - /*
|
| - * Set 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;
|
| -
|
| - /*
|
| - * Get SSRC
|
| - */
|
| - virtual uint32_t SSRC() const = 0;
|
| -
|
| - /*
|
| - * configure SSRC, default is a random number
|
| - */
|
| - virtual void SetSSRC(uint32_t ssrc) = 0;
|
| -
|
| - /*
|
| - * Set CSRC
|
| - *
|
| - * csrcs - vector of CSRCs
|
| - */
|
| - virtual void SetCsrcs(const std::vector<uint32_t>& csrcs) = 0;
|
| -
|
| - /*
|
| - * Turn 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;
|
| -
|
| - /*
|
| - * Get 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;
|
| -
|
| - /*
|
| - * sends kRtcpByeCode when going from true to false
|
| - *
|
| - * sending - on/off
|
| - *
|
| - * return -1 on failure else 0
|
| - */
|
| - virtual int32_t SetSendingStatus(bool sending) = 0;
|
| -
|
| - /*
|
| - * get send status
|
| - */
|
| - virtual bool Sending() const = 0;
|
| -
|
| - /*
|
| - * Starts/Stops media packets, on by default
|
| - *
|
| - * sending - on/off
|
| - */
|
| - virtual void SetSendingMediaStatus(bool sending) = 0;
|
| -
|
| - /*
|
| - * get send status
|
| - */
|
| - virtual bool SendingMedia() const = 0;
|
| -
|
| - /*
|
| - * get sent bitrate in Kbit/s
|
| - */
|
| - virtual void BitrateSent(uint32_t* totalRate,
|
| - uint32_t* videoRate,
|
| - uint32_t* fecRate,
|
| - uint32_t* nackRate) const = 0;
|
| -
|
| - /*
|
| - * Used by the codec module to deliver a video or audio frame for
|
| - * packetization.
|
| - *
|
| - * frameType - type of frame to send
|
| - * payloadType - payload type of frame to send
|
| - * timestamp - timestamp of frame to send
|
| - * payloadData - payload buffer of frame to send
|
| - * payloadSize - size of payload buffer to send
|
| - * fragmentation - fragmentation offset data for fragmented frames such
|
| - * as layers or RED
|
| - *
|
| - * return -1 on failure else 0
|
| - */
|
| - virtual int32_t SendOutgoingData(
|
| - FrameType frameType,
|
| - int8_t payloadType,
|
| - uint32_t timeStamp,
|
| - int64_t capture_time_ms,
|
| - const uint8_t* payloadData,
|
| - size_t payloadSize,
|
| - const RTPFragmentationHeader* fragmentation = NULL,
|
| - const RTPVideoHeader* rtpVideoHdr = NULL) = 0;
|
| -
|
| - 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
|
| - *
|
| - ***************************************************************************/
|
| -
|
| - /*
|
| - * Get RTCP status
|
| - */
|
| - virtual RtcpMode RTCP() const = 0;
|
| -
|
| - /*
|
| - * configure RTCP status i.e on(compound or non- compound)/off
|
| - *
|
| - * method - RTCP method to use
|
| - */
|
| - virtual void SetRTCPStatus(RtcpMode method) = 0;
|
| -
|
| - /*
|
| - * Set RTCP CName (i.e unique identifier)
|
| - *
|
| - * return -1 on failure else 0
|
| - */
|
| - virtual int32_t SetCNAME(const char* c_name) = 0;
|
| -
|
| - /*
|
| - * Get remote CName
|
| - *
|
| - * return -1 on failure else 0
|
| - */
|
| - virtual int32_t RemoteCNAME(uint32_t remoteSSRC,
|
| - char cName[RTCP_CNAME_SIZE]) const = 0;
|
| -
|
| - /*
|
| - * Get remote NTP
|
| - *
|
| - * return -1 on failure else 0
|
| - */
|
| - virtual int32_t RemoteNTP(
|
| - uint32_t *ReceivedNTPsecs,
|
| - uint32_t *ReceivedNTPfrac,
|
| - uint32_t *RTCPArrivalTimeSecs,
|
| - uint32_t *RTCPArrivalTimeFrac,
|
| - uint32_t *rtcp_timestamp) const = 0;
|
| -
|
| - /*
|
| - * AddMixedCNAME
|
| - *
|
| - * return -1 on failure else 0
|
| - */
|
| - virtual int32_t AddMixedCNAME(uint32_t SSRC, const char* c_name) = 0;
|
| -
|
| - /*
|
| - * RemoveMixedCNAME
|
| - *
|
| - * return -1 on failure else 0
|
| - */
|
| - virtual int32_t RemoveMixedCNAME(uint32_t SSRC) = 0;
|
| -
|
| - /*
|
| - * Get RoundTripTime
|
| - *
|
| - * return -1 on failure else 0
|
| - */
|
| - virtual int32_t RTT(uint32_t remoteSSRC,
|
| - int64_t* RTT,
|
| - int64_t* avgRTT,
|
| - int64_t* minRTT,
|
| - int64_t* maxRTT) const = 0;
|
| -
|
| - /*
|
| - * Force a send of a RTCP packet
|
| - * periodic SR and RR are triggered via the process function
|
| - *
|
| - * return -1 on failure else 0
|
| - */
|
| - virtual int32_t SendRTCP(RTCPPacketType rtcpPacketType) = 0;
|
| -
|
| - /*
|
| - * Force a send of a RTCP packet with more than one packet type.
|
| - * periodic SR and RR are triggered via the process function
|
| - *
|
| - * return -1 on failure else 0
|
| - */
|
| - virtual int32_t SendCompoundRTCP(
|
| - const std::set<RTCPPacketType>& rtcpPacketTypes) = 0;
|
| -
|
| - /*
|
| - * Good state of RTP receiver inform sender
|
| - */
|
| - virtual int32_t SendRTCPReferencePictureSelection(uint64_t pictureID) = 0;
|
| -
|
| - /*
|
| - * Send a RTCP Slice Loss Indication (SLI)
|
| - * 6 least significant bits of pictureID
|
| - */
|
| - virtual int32_t SendRTCPSliceLossIndication(uint8_t pictureID) = 0;
|
| -
|
| - /*
|
| - * Statistics of the amount of data sent
|
| - *
|
| - * return -1 on failure else 0
|
| - */
|
| - virtual int32_t DataCountersRTP(
|
| - size_t* bytesSent,
|
| - uint32_t* packetsSent) const = 0;
|
| -
|
| - /*
|
| - * Get send statistics for the RTP and RTX stream.
|
| - */
|
| - virtual void GetSendStreamDataCounters(
|
| - StreamDataCounters* rtp_counters,
|
| - StreamDataCounters* rtx_counters) const = 0;
|
| -
|
| - /*
|
| - * Get packet loss statistics for the RTP stream.
|
| - */
|
| - virtual void GetRtpPacketLossStats(
|
| - bool outgoing,
|
| - uint32_t ssrc,
|
| - struct RtpPacketLossStats* loss_stats) const = 0;
|
| -
|
| - /*
|
| - * Get received RTCP sender info
|
| - *
|
| - * return -1 on failure else 0
|
| - */
|
| - virtual int32_t RemoteRTCPStat(RTCPSenderInfo* senderInfo) = 0;
|
| -
|
| - /*
|
| - * Get received RTCP report block
|
| - *
|
| - * return -1 on failure else 0
|
| - */
|
| - virtual int32_t RemoteRTCPStat(
|
| - std::vector<RTCPReportBlock>* receiveBlocks) const = 0;
|
| -
|
| - /*
|
| - * (APP) Application specific data
|
| - *
|
| - * return -1 on failure else 0
|
| - */
|
| - virtual int32_t SetRTCPApplicationSpecificData(uint8_t subType,
|
| - uint32_t name,
|
| - const uint8_t* data,
|
| - uint16_t length) = 0;
|
| - /*
|
| - * (XR) VOIP metric
|
| - *
|
| - * return -1 on failure else 0
|
| - */
|
| - virtual int32_t SetRTCPVoIPMetrics(
|
| - const RTCPVoIPMetric* VoIPMetric) = 0;
|
| -
|
| - /*
|
| - * (XR) Receiver Reference Time Report
|
| - */
|
| - virtual void SetRtcpXrRrtrStatus(bool enable) = 0;
|
| -
|
| - 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;
|
| -
|
| - /*
|
| - * Send a Negative acknowledgement packet
|
| - *
|
| - * return -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* nackList, uint16_t size) = 0;
|
| -
|
| - /*
|
| - * Send 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 acknowledgement
|
| - * 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
|
| - *
|
| - ***************************************************************************/
|
| -
|
| - /*
|
| - * set audio packet size, used to determine when it's time to send a DTMF
|
| - * packet in silence (CNG)
|
| - *
|
| - * return -1 on failure else 0
|
| - */
|
| - virtual int32_t SetAudioPacketSize(uint16_t packetSizeSamples) = 0;
|
| -
|
| - /*
|
| - * Send a TelephoneEvent tone using RFC 2833 (4733)
|
| - *
|
| - * return -1 on failure else 0
|
| - */
|
| - virtual int32_t SendTelephoneEventOutband(uint8_t key,
|
| - uint16_t time_ms,
|
| - uint8_t level) = 0;
|
| -
|
| - /*
|
| - * Set payload type for Redundant Audio Data RFC 2198
|
| - *
|
| - * return -1 on failure else 0
|
| - */
|
| - virtual int32_t SetSendREDPayloadType(int8_t payloadType) = 0;
|
| -
|
| - /*
|
| - * Get payload type for Redundant Audio Data RFC 2198
|
| - *
|
| - * return -1 on failure else 0
|
| - */
|
| - virtual int32_t SendREDPayloadType(int8_t* payload_type) const = 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
|
| - *
|
| - * return -1 on failure else 0
|
| - */
|
| - virtual int32_t SetKeyFrameRequestMethod(KeyFrameRequestMethod method) = 0;
|
| -
|
| - /*
|
| - * send a request for a keyframe
|
| - *
|
| - * return -1 on failure else 0
|
| - */
|
| - virtual int32_t RequestKeyFrame() = 0;
|
| + // **************************************************************************
|
| + // 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
|
| + // Returns -1 on failure else 0.
|
| + virtual 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) = 0;
|
| +
|
| + 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;
|
| +
|
| + // Sets payload type for Redundant Audio Data RFC 2198.
|
| + // Returns -1 on failure else 0.
|
| + virtual int32_t SetSendREDPayloadType(int8_t payload_type) = 0;
|
| +
|
| + // Get payload type for Redundant Audio Data RFC 2198.
|
| + // Returns -1 on failure else 0.
|
| + virtual int32_t SendREDPayloadType(int8_t* payload_type) const = 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_
|
|
|
Chromium Code Reviews
Issue 2067673004:
Style cleanups in RtpSender. (Closed)
Base URL: https://chromium.googlesource.com/external/webrtc.git@master