modules/rtp_rtcp/include/rtp_rtcp.h - src.git - Git at Google

 /*
  *  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_INCLUDE_RTP_RTCP_H_
 #define MODULES_RTP_RTCP_INCLUDE_RTP_RTCP_H_

 #include <set>
 #include <string>
 #include <utility>
 #include <vector>

 #include "absl/types/optional.h"
 #include "api/video/video_bitrate_allocation.h"
 #include "common_types.h"  // NOLINT(build/include)
 #include "modules/include/module.h"
 #include "modules/rtp_rtcp/include/flexfec_sender.h"
 #include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
 #include "rtc_base/constructormagic.h"
 #include "rtc_base/deprecation.h"

 namespace webrtc {

 // Forward declarations.
 class OverheadObserver;
 class RateLimiter;
 class ReceiveStatisticsProvider;
 class RemoteBitrateEstimator;
 class RtcEventLog;
 class Transport;
 class VideoBitrateAllocationObserver;

 namespace rtcp {
 class TransportFeedback;
 }

 class RtpRtcp : public Module, public RtcpFeedbackSenderInterface {
  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;

     ReceiveStatisticsProvider* receive_statistics = nullptr;

     // 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;
     VideoBitrateAllocationObserver* bitrate_allocation_observer = 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;

     // Generate FlexFEC packets.
     // TODO(brandtr): Remove when FlexfecSender is wired up to PacedSender.
     FlexfecSender* flexfec_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;
     OverheadObserver* overhead_observer = nullptr;
     RtpKeepAliveConfig keepalive_config;
     RtcpIntervalConfig rtcp_interval_config;

     // Update network2 instead of pacer_exit field of video timing extension.
     bool populate_network2_timestamp = false;

    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 void IncomingRtcpPacket(const uint8_t* incoming_packet,
                                   size_t incoming_packet_length) = 0;

   virtual void SetRemoteSSRC(uint32_t ssrc) = 0;

   // **************************************************************************
   // Sender
   // **************************************************************************

   // Sets the maximum size of an RTP packet, including RTP headers.
   virtual void SetMaxRtpPacketSize(size_t size) = 0;

   // Returns max RTP packet size. Takes into account RTP headers and
   // FEC/ULP/RED overhead (when FEC is enabled).
   virtual size_t MaxRtpPacketSize() const = 0;

   // Sets codec name and payload type. Returns -1 on failure else 0.
   virtual int32_t RegisterSendPayload(const CodecInst& voice_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;
   // Register extension by uri, returns false on failure.
   virtual bool RegisterRtpHeaderExtension(const std::string& uri, int id) = 0;

   virtual int32_t DeregisterSendRtpHeaderExtension(RTPExtensionType type) = 0;

   virtual bool HasBweExtensions() const = 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.
   uint32_t SSRC() const override = 0;

   // Sets SSRC, default is a random number.
   virtual void SetSSRC(uint32_t ssrc) = 0;

   // Sets the value for sending in the MID RTP header extension.
   // The MID RTP header extension should be registered for this to do anything.
   // Once set, this value can not be changed or removed.
   virtual void SetMid(const std::string& mid) = 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;

   // Returns the FlexFEC SSRC, if there is one.
   virtual absl::optional<uint32_t> FlexfecSsrc() const = 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;

   virtual bool TimeToSendPacket(uint32_t ssrc,
                                 uint16_t sequence_number,
                                 int64_t capture_time_ms,
                                 bool retransmission,
                                 const PacedPacketInfo& pacing_info) = 0;

   virtual size_t TimeToSendPadding(size_t bytes,
                                    const PacedPacketInfo& pacing_info) = 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;

   // 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 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.
   // Schedules sending REMB on next and following sender/receiver reports.
   void SetRemb(int64_t bitrate_bps, std::vector<uint32_t> ssrcs) override = 0;
   // Stops sending REMB on next and following sender/receiver reports.
   void UnsetRemb() override = 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.
   bool SendFeedbackPacket(const rtcp::TransportFeedback& packet) override = 0;

   virtual void SetVideoBitrateAllocation(
       const VideoBitrateAllocation& bitrate) = 0;

   // **************************************************************************
   // Audio
   // **************************************************************************

   // 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
   // **************************************************************************

   // Set RED and ULPFEC payload types. A payload type of -1 means that the
   // corresponding feature is turned off. Note that we DO NOT support enabling
   // ULPFEC without enabling RED, and RED is only ever used when ULPFEC is
   // enabled.
   virtual void SetUlpfecConfig(int red_payload_type,
                                int ulpfec_payload_type) = 0;

   // Set FEC rates, max frames before FEC is sent, and type of FEC masks.
   // Returns false on failure.
   virtual bool SetFecParameters(const FecProtectionParams& delta_params,
                                 const FecProtectionParams& key_params) = 0;

   // Deprecated version of member function above.
   RTC_DEPRECATED
   int32_t SetFecParameters(const FecProtectionParams* delta_params,
                            const FecProtectionParams* key_params);

   // 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  // MODULES_RTP_RTCP_INCLUDE_RTP_RTCP_H_
	/*
	* 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_INCLUDE_RTP_RTCP_H_
	#define MODULES_RTP_RTCP_INCLUDE_RTP_RTCP_H_

	#include <set>
	#include <string>
	#include <utility>
	#include <vector>

	#include "absl/types/optional.h"
	#include "api/video/video_bitrate_allocation.h"
	#include "common_types.h" // NOLINT(build/include)
	#include "modules/include/module.h"
	#include "modules/rtp_rtcp/include/flexfec_sender.h"
	#include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
	#include "rtc_base/constructormagic.h"
	#include "rtc_base/deprecation.h"

	namespace webrtc {

	// Forward declarations.
	class OverheadObserver;
	class RateLimiter;
	class ReceiveStatisticsProvider;
	class RemoteBitrateEstimator;
	class RtcEventLog;
	class Transport;
	class VideoBitrateAllocationObserver;

	namespace rtcp {
	class TransportFeedback;
	}

	class RtpRtcp : public Module, public RtcpFeedbackSenderInterface {
	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;

	ReceiveStatisticsProvider* receive_statistics = nullptr;

	// 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;
	VideoBitrateAllocationObserver* bitrate_allocation_observer = 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;

	// Generate FlexFEC packets.
	// TODO(brandtr): Remove when FlexfecSender is wired up to PacedSender.
	FlexfecSender* flexfec_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;
	OverheadObserver* overhead_observer = nullptr;
	RtpKeepAliveConfig keepalive_config;
	RtcpIntervalConfig rtcp_interval_config;

	// Update network2 instead of pacer_exit field of video timing extension.
	bool populate_network2_timestamp = false;

	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 void IncomingRtcpPacket(const uint8_t* incoming_packet,
	size_t incoming_packet_length) = 0;

	virtual void SetRemoteSSRC(uint32_t ssrc) = 0;

	// **************************************************************************
	// Sender
	// **************************************************************************

	// Sets the maximum size of an RTP packet, including RTP headers.
	virtual void SetMaxRtpPacketSize(size_t size) = 0;

	// Returns max RTP packet size. Takes into account RTP headers and
	// FEC/ULP/RED overhead (when FEC is enabled).
	virtual size_t MaxRtpPacketSize() const = 0;

	// Sets codec name and payload type. Returns -1 on failure else 0.
	virtual int32_t RegisterSendPayload(const CodecInst& voice_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;
	// Register extension by uri, returns false on failure.
	virtual bool RegisterRtpHeaderExtension(const std::string& uri, int id) = 0;

	virtual int32_t DeregisterSendRtpHeaderExtension(RTPExtensionType type) = 0;

	virtual bool HasBweExtensions() const = 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.
	uint32_t SSRC() const override = 0;

	// Sets SSRC, default is a random number.
	virtual void SetSSRC(uint32_t ssrc) = 0;

	// Sets the value for sending in the MID RTP header extension.
	// The MID RTP header extension should be registered for this to do anything.
	// Once set, this value can not be changed or removed.
	virtual void SetMid(const std::string& mid) = 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;

	// Returns the FlexFEC SSRC, if there is one.
	virtual absl::optional<uint32_t> FlexfecSsrc() const = 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;

	virtual bool TimeToSendPacket(uint32_t ssrc,
	uint16_t sequence_number,
	int64_t capture_time_ms,
	bool retransmission,
	const PacedPacketInfo& pacing_info) = 0;

	virtual size_t TimeToSendPadding(size_t bytes,
	const PacedPacketInfo& pacing_info) = 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;

	// 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 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.
	// Schedules sending REMB on next and following sender/receiver reports.
	void SetRemb(int64_t bitrate_bps, std::vector<uint32_t> ssrcs) override = 0;
	// Stops sending REMB on next and following sender/receiver reports.
	void UnsetRemb() override = 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.
	bool SendFeedbackPacket(const rtcp::TransportFeedback& packet) override = 0;

	virtual void SetVideoBitrateAllocation(
	const VideoBitrateAllocation& bitrate) = 0;

	// **************************************************************************
	// Audio
	// **************************************************************************

	// 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
	// **************************************************************************

	// Set RED and ULPFEC payload types. A payload type of -1 means that the
	// corresponding feature is turned off. Note that we DO NOT support enabling
	// ULPFEC without enabling RED, and RED is only ever used when ULPFEC is
	// enabled.
	virtual void SetUlpfecConfig(int red_payload_type,
	int ulpfec_payload_type) = 0;

	// Set FEC rates, max frames before FEC is sent, and type of FEC masks.
	// Returns false on failure.
	virtual bool SetFecParameters(const FecProtectionParams& delta_params,
	const FecProtectionParams& key_params) = 0;

	// Deprecated version of member function above.
	RTC_DEPRECATED
	int32_t SetFecParameters(const FecProtectionParams* delta_params,
	const FecProtectionParams* key_params);

	// 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 // MODULES_RTP_RTCP_INCLUDE_RTP_RTCP_H_