modules/rtp_rtcp/include/rtp_rtcp_defines.h - src - 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_DEFINES_H_
 #define MODULES_RTP_RTCP_INCLUDE_RTP_RTCP_DEFINES_H_

 #include <stddef.h>
 #include <list>
 #include <vector>

 #include "api/audio_codecs/audio_format.h"
 #include "api/rtp_headers.h"
 #include "common_types.h"  // NOLINT(build/include)
 #include "modules/include/module_common_types.h"
 #include "system_wrappers/include/clock.h"
 #include "typedefs.h"  // NOLINT(build/include)

 #define RTCP_CNAME_SIZE 256  // RFC 3550 page 44, including null termination
 #define IP_PACKET_SIZE 1500  // we assume ethernet
 #define MAX_NUMBER_OF_PARALLEL_TELEPHONE_EVENTS 10

 namespace webrtc {
 namespace rtcp {
 class TransportFeedback;
 }

 const int kVideoPayloadTypeFrequency = 90000;
 // TODO(solenberg): RTP time stamp rate for RTCP is fixed at 8k, this is legacy
 // and should be fixed.
 // See: https://bugs.chromium.org/p/webrtc/issues/detail?id=6458
 const int kBogusRtpRateForAudioRtcp = 8000;

 // Minimum RTP header size in bytes.
 const uint8_t kRtpHeaderSize = 12;

 struct RtcpIntervalConfig final {
   RtcpIntervalConfig() = default;
   RtcpIntervalConfig(int64_t video_interval_ms, int64_t audio_interval_ms)
       : video_interval_ms(video_interval_ms),
         audio_interval_ms(audio_interval_ms) {}
   int64_t video_interval_ms = 1000;
   int64_t audio_interval_ms = 5000;
 };

 struct AudioPayload {
   SdpAudioFormat format;
   uint32_t rate;
 };

 struct VideoPayload {
   VideoCodecType videoCodecType;
   // The H264 profile only matters if videoCodecType == kVideoCodecH264.
   H264::Profile h264_profile;
 };

 class PayloadUnion {
  public:
   explicit PayloadUnion(const AudioPayload& payload);
   explicit PayloadUnion(const VideoPayload& payload);
   PayloadUnion(const PayloadUnion&);
   PayloadUnion(PayloadUnion&&);
   ~PayloadUnion();

   PayloadUnion& operator=(const PayloadUnion&);
   PayloadUnion& operator=(PayloadUnion&&);

   bool is_audio() const { return audio_payload_.has_value(); }
   bool is_video() const { return video_payload_.has_value(); }
   const AudioPayload& audio_payload() const {
     RTC_DCHECK(audio_payload_);
     return *audio_payload_;
   }
   const VideoPayload& video_payload() const {
     RTC_DCHECK(video_payload_);
     return *video_payload_;
   }
   AudioPayload& audio_payload() {
     RTC_DCHECK(audio_payload_);
     return *audio_payload_;
   }
   VideoPayload& video_payload() {
     RTC_DCHECK(video_payload_);
     return *video_payload_;
   }

  private:
   absl::optional<AudioPayload> audio_payload_;
   absl::optional<VideoPayload> video_payload_;
 };

 enum RTPAliveType { kRtpDead = 0, kRtpNoRtp = 1, kRtpAlive = 2 };

 enum ProtectionType { kUnprotectedPacket, kProtectedPacket };

 enum StorageType { kDontRetransmit, kAllowRetransmission };

 enum RTPExtensionType {
   kRtpExtensionNone,
   kRtpExtensionTransmissionTimeOffset,
   kRtpExtensionAudioLevel,
   kRtpExtensionAbsoluteSendTime,
   kRtpExtensionVideoRotation,
   kRtpExtensionTransportSequenceNumber,
   kRtpExtensionPlayoutDelay,
   kRtpExtensionVideoContentType,
   kRtpExtensionVideoTiming,
   kRtpExtensionRtpStreamId,
   kRtpExtensionRepairedRtpStreamId,
   kRtpExtensionMid,
   kRtpExtensionNumberOfExtensions  // Must be the last entity in the enum.
 };

 enum RTCPAppSubTypes { kAppSubtypeBwe = 0x00 };

 // TODO(sprang): Make this an enum class once rtcp_receiver has been cleaned up.
 enum RTCPPacketType : uint32_t {
   kRtcpReport = 0x0001,
   kRtcpSr = 0x0002,
   kRtcpRr = 0x0004,
   kRtcpSdes = 0x0008,
   kRtcpBye = 0x0010,
   kRtcpPli = 0x0020,
   kRtcpNack = 0x0040,
   kRtcpFir = 0x0080,
   kRtcpTmmbr = 0x0100,
   kRtcpTmmbn = 0x0200,
   kRtcpSrReq = 0x0400,
   kRtcpXrVoipMetric = 0x0800,
   kRtcpApp = 0x1000,
   kRtcpRemb = 0x10000,
   kRtcpTransmissionTimeOffset = 0x20000,
   kRtcpXrReceiverReferenceTime = 0x40000,
   kRtcpXrDlrrReportBlock = 0x80000,
   kRtcpTransportFeedback = 0x100000,
   kRtcpXrTargetBitrate = 0x200000
 };

 enum KeyFrameRequestMethod { kKeyFrameReqPliRtcp, kKeyFrameReqFirRtcp };

 enum RtpRtcpPacketType { kPacketRtp = 0, kPacketKeepAlive = 1 };

 // kConditionallyRetransmitHigherLayers allows retransmission of video frames
 // in higher layers if either the last frame in that layer was too far back in
 // time, or if we estimate that a new frame will be available in a lower layer
 // in a shorter time than it would take to request and receive a retransmission.
 enum RetransmissionMode : uint8_t {
   kRetransmitOff = 0x0,
   kRetransmitFECPackets = 0x1,
   kRetransmitBaseLayer = 0x2,
   kRetransmitHigherLayers = 0x4,
   kConditionallyRetransmitHigherLayers = 0x8,
   kRetransmitAllPackets = 0xFF
 };

 enum RtxMode {
   kRtxOff = 0x0,
   kRtxRetransmitted = 0x1,     // Only send retransmissions over RTX.
   kRtxRedundantPayloads = 0x2  // Preventively send redundant payloads
                                // instead of padding.
 };

 const size_t kRtxHeaderSize = 2;

 struct RTCPReportBlock {
   RTCPReportBlock()
       : sender_ssrc(0),
         source_ssrc(0),
         fraction_lost(0),
         packets_lost(0),
         extended_highest_sequence_number(0),
         jitter(0),
         last_sender_report_timestamp(0),
         delay_since_last_sender_report(0) {}

   RTCPReportBlock(uint32_t sender_ssrc,
                   uint32_t source_ssrc,
                   uint8_t fraction_lost,
                   int32_t packets_lost,
                   uint32_t extended_highest_sequence_number,
                   uint32_t jitter,
                   uint32_t last_sender_report_timestamp,
                   uint32_t delay_since_last_sender_report)
       : sender_ssrc(sender_ssrc),
         source_ssrc(source_ssrc),
         fraction_lost(fraction_lost),
         packets_lost(packets_lost),
         extended_highest_sequence_number(extended_highest_sequence_number),
         jitter(jitter),
         last_sender_report_timestamp(last_sender_report_timestamp),
         delay_since_last_sender_report(delay_since_last_sender_report) {}

   // Fields as described by RFC 3550 6.4.2.
   uint32_t sender_ssrc;  // SSRC of sender of this report.
   uint32_t source_ssrc;  // SSRC of the RTP packet sender.
   uint8_t fraction_lost;
   uint32_t packets_lost;  // 24 bits valid.
   uint32_t extended_highest_sequence_number;
   uint32_t jitter;
   uint32_t last_sender_report_timestamp;
   uint32_t delay_since_last_sender_report;
 };

 typedef std::list<RTCPReportBlock> ReportBlockList;

 struct RtpState {
   RtpState()
       : sequence_number(0),
         start_timestamp(0),
         timestamp(0),
         capture_time_ms(-1),
         last_timestamp_time_ms(-1),
         media_has_been_sent(false) {}
   uint16_t sequence_number;
   uint32_t start_timestamp;
   uint32_t timestamp;
   int64_t capture_time_ms;
   int64_t last_timestamp_time_ms;
   bool media_has_been_sent;
 };

 class RtpData {
  public:
   virtual ~RtpData() {}

   virtual int32_t OnReceivedPayloadData(const uint8_t* payload_data,
                                         size_t payload_size,
                                         const WebRtcRTPHeader* rtp_header) = 0;
 };

 // Callback interface for packets recovered by FlexFEC or ULPFEC. In
 // the FlexFEC case, the implementation should be able to demultiplex
 // the recovered RTP packets based on SSRC.
 class RecoveredPacketReceiver {
  public:
   virtual void OnRecoveredPacket(const uint8_t* packet, size_t length) = 0;

  protected:
   virtual ~RecoveredPacketReceiver() = default;
 };

 class RtcpIntraFrameObserver {
  public:
   virtual ~RtcpIntraFrameObserver() {}

   virtual void OnReceivedIntraFrameRequest(uint32_t ssrc) = 0;
 };

 class RtcpBandwidthObserver {
  public:
   // REMB or TMMBR
   virtual void OnReceivedEstimatedBitrate(uint32_t bitrate) = 0;

   virtual void OnReceivedRtcpReceiverReport(
       const ReportBlockList& report_blocks,
       int64_t rtt,
       int64_t now_ms) = 0;

   virtual ~RtcpBandwidthObserver() {}
 };

 struct PacketFeedback {
   PacketFeedback(int64_t arrival_time_ms, uint16_t sequence_number)
       : PacketFeedback(-1,
                        arrival_time_ms,
                        kNoSendTime,
                        sequence_number,
                        0,
                        0,
                        0,
                        PacedPacketInfo()) {}

   PacketFeedback(int64_t arrival_time_ms,
                  int64_t send_time_ms,
                  uint16_t sequence_number,
                  size_t payload_size,
                  const PacedPacketInfo& pacing_info)
       : PacketFeedback(-1,
                        arrival_time_ms,
                        send_time_ms,
                        sequence_number,
                        payload_size,
                        0,
                        0,
                        pacing_info) {}

   PacketFeedback(int64_t creation_time_ms,
                  uint16_t sequence_number,
                  size_t payload_size,
                  uint16_t local_net_id,
                  uint16_t remote_net_id,
                  const PacedPacketInfo& pacing_info)
       : PacketFeedback(creation_time_ms,
                        kNotReceived,
                        kNoSendTime,
                        sequence_number,
                        payload_size,
                        local_net_id,
                        remote_net_id,
                        pacing_info) {}

   PacketFeedback(int64_t creation_time_ms,
                  int64_t arrival_time_ms,
                  int64_t send_time_ms,
                  uint16_t sequence_number,
                  size_t payload_size,
                  uint16_t local_net_id,
                  uint16_t remote_net_id,
                  const PacedPacketInfo& pacing_info)
       : creation_time_ms(creation_time_ms),
         arrival_time_ms(arrival_time_ms),
         send_time_ms(send_time_ms),
         sequence_number(sequence_number),
         payload_size(payload_size),
         local_net_id(local_net_id),
         remote_net_id(remote_net_id),
         pacing_info(pacing_info) {}

   static constexpr int kNotAProbe = -1;
   static constexpr int64_t kNotReceived = -1;
   static constexpr int64_t kNoSendTime = -1;

   // NOTE! The variable |creation_time_ms| is not used when testing equality.
   //       This is due to |creation_time_ms| only being used by SendTimeHistory
   //       for book-keeping, and is of no interest outside that class.
   // TODO(philipel): Remove |creation_time_ms| from PacketFeedback when cleaning
   //                 up SendTimeHistory.
   bool operator==(const PacketFeedback& rhs) const {
     return arrival_time_ms == rhs.arrival_time_ms &&
            send_time_ms == rhs.send_time_ms &&
            sequence_number == rhs.sequence_number &&
            payload_size == rhs.payload_size && pacing_info == rhs.pacing_info;
   }

   // Time corresponding to when this object was created.
   int64_t creation_time_ms;
   // Time corresponding to when the packet was received. Timestamped with the
   // receiver's clock. For unreceived packet, the sentinel value kNotReceived
   // is used.
   int64_t arrival_time_ms;
   // Time corresponding to when the packet was sent, timestamped with the
   // sender's clock.
   int64_t send_time_ms;
   // Packet identifier, incremented with 1 for every packet generated by the
   // sender.
   uint16_t sequence_number;
   // Session unique packet identifier, incremented with 1 for every packet
   // generated by the sender.
   int64_t long_sequence_number;
   // Size of the packet excluding RTP headers.
   size_t payload_size;
   // The network route ids that this packet is associated with.
   uint16_t local_net_id;
   uint16_t remote_net_id;
   // Pacing information about this packet.
   PacedPacketInfo pacing_info;
 };

 class PacketFeedbackComparator {
  public:
   inline bool operator()(const PacketFeedback& lhs, const PacketFeedback& rhs) {
     if (lhs.arrival_time_ms != rhs.arrival_time_ms)
       return lhs.arrival_time_ms < rhs.arrival_time_ms;
     if (lhs.send_time_ms != rhs.send_time_ms)
       return lhs.send_time_ms < rhs.send_time_ms;
     return lhs.sequence_number < rhs.sequence_number;
   }
 };

 class TransportFeedbackObserver {
  public:
   TransportFeedbackObserver() {}
   virtual ~TransportFeedbackObserver() {}

   // Note: Transport-wide sequence number as sequence number.
   virtual void AddPacket(uint32_t ssrc,
                          uint16_t sequence_number,
                          size_t length,
                          const PacedPacketInfo& pacing_info) = 0;

   virtual void OnTransportFeedback(const rtcp::TransportFeedback& feedback) = 0;
 };

 // Interface for PacketRouter to send rtcp feedback on behalf of
 // congestion controller.
 // TODO(bugs.webrtc.org/8239): Remove and use RtcpTransceiver directly
 // when RtcpTransceiver always present in rtp transport.
 class RtcpFeedbackSenderInterface {
  public:
   virtual ~RtcpFeedbackSenderInterface() = default;
   virtual uint32_t SSRC() const = 0;
   virtual bool SendFeedbackPacket(const rtcp::TransportFeedback& feedback) = 0;
   virtual void SetRemb(int64_t bitrate_bps, std::vector<uint32_t> ssrcs) = 0;
   virtual void UnsetRemb() = 0;
 };

 class PacketFeedbackObserver {
  public:
   virtual ~PacketFeedbackObserver() = default;

   virtual void OnPacketAdded(uint32_t ssrc, uint16_t seq_num) = 0;
   virtual void OnPacketFeedbackVector(
       const std::vector<PacketFeedback>& packet_feedback_vector) = 0;
 };

 class RtcpRttStats {
  public:
   virtual void OnRttUpdate(int64_t rtt) = 0;

   virtual int64_t LastProcessedRtt() const = 0;

   virtual ~RtcpRttStats() {}
 };

 // Statistics about packet loss for a single directional connection. All values
 // are totals since the connection initiated.
 struct RtpPacketLossStats {
   // The number of packets lost in events where no adjacent packets were also
   // lost.
   uint64_t single_packet_loss_count;
   // The number of events in which more than one adjacent packet was lost.
   uint64_t multiple_packet_loss_event_count;
   // The number of packets lost in events where more than one adjacent packet
   // was lost.
   uint64_t multiple_packet_loss_packet_count;
 };

 class RtpPacketSender {
  public:
   RtpPacketSender() {}
   virtual ~RtpPacketSender() {}

   enum Priority {
     kHighPriority = 0,    // Pass through; will be sent immediately.
     kNormalPriority = 2,  // Put in back of the line.
     kLowPriority = 3,     // Put in back of the low priority line.
   };
   // Low priority packets are mixed with the normal priority packets
   // while we are paused.

   // Returns true if we send the packet now, else it will add the packet
   // information to the queue and call TimeToSendPacket when it's time to send.
   virtual void InsertPacket(Priority priority,
                             uint32_t ssrc,
                             uint16_t sequence_number,
                             int64_t capture_time_ms,
                             size_t bytes,
                             bool retransmission) = 0;

   // Currently audio traffic is not accounted by pacer and passed through.
   // With the introduction of audio BWE audio traffic will be accounted for
   // the pacer budget calculation. The audio traffic still will be injected
   // at high priority.
   // TODO(alexnarest): Make it pure virtual after rtp_sender_unittest will be
   // updated to support it
   virtual void SetAccountForAudioPackets(bool account_for_audio) {}
 };

 class TransportSequenceNumberAllocator {
  public:
   TransportSequenceNumberAllocator() {}
   virtual ~TransportSequenceNumberAllocator() {}

   virtual uint16_t AllocateSequenceNumber() = 0;
 };

 struct RtpPacketCounter {
   RtpPacketCounter()
       : header_bytes(0), payload_bytes(0), padding_bytes(0), packets(0) {}

   void Add(const RtpPacketCounter& other) {
     header_bytes += other.header_bytes;
     payload_bytes += other.payload_bytes;
     padding_bytes += other.padding_bytes;
     packets += other.packets;
   }

   void Subtract(const RtpPacketCounter& other) {
     RTC_DCHECK_GE(header_bytes, other.header_bytes);
     header_bytes -= other.header_bytes;
     RTC_DCHECK_GE(payload_bytes, other.payload_bytes);
     payload_bytes -= other.payload_bytes;
     RTC_DCHECK_GE(padding_bytes, other.padding_bytes);
     padding_bytes -= other.padding_bytes;
     RTC_DCHECK_GE(packets, other.packets);
     packets -= other.packets;
   }

   void AddPacket(size_t packet_length, const RTPHeader& header) {
     ++packets;
     header_bytes += header.headerLength;
     padding_bytes += header.paddingLength;
     payload_bytes +=
         packet_length - (header.headerLength + header.paddingLength);
   }

   size_t TotalBytes() const {
     return header_bytes + payload_bytes + padding_bytes;
   }

   size_t header_bytes;   // Number of bytes used by RTP headers.
   size_t payload_bytes;  // Payload bytes, excluding RTP headers and padding.
   size_t padding_bytes;  // Number of padding bytes.
   uint32_t packets;      // Number of packets.
 };

 // Data usage statistics for a (rtp) stream.
 struct StreamDataCounters {
   StreamDataCounters();

   void Add(const StreamDataCounters& other) {
     transmitted.Add(other.transmitted);
     retransmitted.Add(other.retransmitted);
     fec.Add(other.fec);
     if (other.first_packet_time_ms != -1 &&
         (other.first_packet_time_ms < first_packet_time_ms ||
          first_packet_time_ms == -1)) {
       // Use oldest time.
       first_packet_time_ms = other.first_packet_time_ms;
     }
   }

   void Subtract(const StreamDataCounters& other) {
     transmitted.Subtract(other.transmitted);
     retransmitted.Subtract(other.retransmitted);
     fec.Subtract(other.fec);
     if (other.first_packet_time_ms != -1 &&
         (other.first_packet_time_ms > first_packet_time_ms ||
          first_packet_time_ms == -1)) {
       // Use youngest time.
       first_packet_time_ms = other.first_packet_time_ms;
     }
   }

   int64_t TimeSinceFirstPacketInMs(int64_t now_ms) const {
     return (first_packet_time_ms == -1) ? -1 : (now_ms - first_packet_time_ms);
   }

   // Returns the number of bytes corresponding to the actual media payload (i.e.
   // RTP headers, padding, retransmissions and fec packets are excluded).
   // Note this function does not have meaning for an RTX stream.
   size_t MediaPayloadBytes() const {
     return transmitted.payload_bytes - retransmitted.payload_bytes -
            fec.payload_bytes;
   }

   int64_t first_packet_time_ms;    // Time when first packet is sent/received.
   RtpPacketCounter transmitted;    // Number of transmitted packets/bytes.
   RtpPacketCounter retransmitted;  // Number of retransmitted packets/bytes.
   RtpPacketCounter fec;            // Number of redundancy packets/bytes.
 };

 // Callback, called whenever byte/packet counts have been updated.
 class StreamDataCountersCallback {
  public:
   virtual ~StreamDataCountersCallback() {}

   virtual void DataCountersUpdated(const StreamDataCounters& counters,
                                    uint32_t ssrc) = 0;
 };

 }  // namespace webrtc
 #endif  // MODULES_RTP_RTCP_INCLUDE_RTP_RTCP_DEFINES_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_DEFINES_H_
	#define MODULES_RTP_RTCP_INCLUDE_RTP_RTCP_DEFINES_H_

	#include <stddef.h>
	#include <list>
	#include <vector>

	#include "api/audio_codecs/audio_format.h"
	#include "api/rtp_headers.h"
	#include "common_types.h" // NOLINT(build/include)
	#include "modules/include/module_common_types.h"
	#include "system_wrappers/include/clock.h"
	#include "typedefs.h" // NOLINT(build/include)

	#define RTCP_CNAME_SIZE 256 // RFC 3550 page 44, including null termination
	#define IP_PACKET_SIZE 1500 // we assume ethernet
	#define MAX_NUMBER_OF_PARALLEL_TELEPHONE_EVENTS 10

	namespace webrtc {
	namespace rtcp {
	class TransportFeedback;
	}

	const int kVideoPayloadTypeFrequency = 90000;
	// TODO(solenberg): RTP time stamp rate for RTCP is fixed at 8k, this is legacy
	// and should be fixed.
	// See: https://bugs.chromium.org/p/webrtc/issues/detail?id=6458
	const int kBogusRtpRateForAudioRtcp = 8000;

	// Minimum RTP header size in bytes.
	const uint8_t kRtpHeaderSize = 12;

	struct RtcpIntervalConfig final {
	RtcpIntervalConfig() = default;
	RtcpIntervalConfig(int64_t video_interval_ms, int64_t audio_interval_ms)
	: video_interval_ms(video_interval_ms),
	audio_interval_ms(audio_interval_ms) {}
	int64_t video_interval_ms = 1000;
	int64_t audio_interval_ms = 5000;
	};

	struct AudioPayload {
	SdpAudioFormat format;
	uint32_t rate;
	};

	struct VideoPayload {
	VideoCodecType videoCodecType;
	// The H264 profile only matters if videoCodecType == kVideoCodecH264.
	H264::Profile h264_profile;
	};

	class PayloadUnion {
	public:
	explicit PayloadUnion(const AudioPayload& payload);
	explicit PayloadUnion(const VideoPayload& payload);
	PayloadUnion(const PayloadUnion&);
	PayloadUnion(PayloadUnion&&);
	~PayloadUnion();

	PayloadUnion& operator=(const PayloadUnion&);
	PayloadUnion& operator=(PayloadUnion&&);

	bool is_audio() const { return audio_payload_.has_value(); }
	bool is_video() const { return video_payload_.has_value(); }
	const AudioPayload& audio_payload() const {
	RTC_DCHECK(audio_payload_);
	return *audio_payload_;
	}
	const VideoPayload& video_payload() const {
	RTC_DCHECK(video_payload_);
	return *video_payload_;
	}
	AudioPayload& audio_payload() {
	RTC_DCHECK(audio_payload_);
	return *audio_payload_;
	}
	VideoPayload& video_payload() {
	RTC_DCHECK(video_payload_);
	return *video_payload_;
	}

	private:
	absl::optional<AudioPayload> audio_payload_;
	absl::optional<VideoPayload> video_payload_;
	};

	enum RTPAliveType { kRtpDead = 0, kRtpNoRtp = 1, kRtpAlive = 2 };

	enum ProtectionType { kUnprotectedPacket, kProtectedPacket };

	enum StorageType { kDontRetransmit, kAllowRetransmission };

	enum RTPExtensionType {
	kRtpExtensionNone,
	kRtpExtensionTransmissionTimeOffset,
	kRtpExtensionAudioLevel,
	kRtpExtensionAbsoluteSendTime,
	kRtpExtensionVideoRotation,
	kRtpExtensionTransportSequenceNumber,
	kRtpExtensionPlayoutDelay,
	kRtpExtensionVideoContentType,
	kRtpExtensionVideoTiming,
	kRtpExtensionRtpStreamId,
	kRtpExtensionRepairedRtpStreamId,
	kRtpExtensionMid,
	kRtpExtensionNumberOfExtensions // Must be the last entity in the enum.
	};

	enum RTCPAppSubTypes { kAppSubtypeBwe = 0x00 };

	// TODO(sprang): Make this an enum class once rtcp_receiver has been cleaned up.
	enum RTCPPacketType : uint32_t {
	kRtcpReport = 0x0001,
	kRtcpSr = 0x0002,
	kRtcpRr = 0x0004,
	kRtcpSdes = 0x0008,
	kRtcpBye = 0x0010,
	kRtcpPli = 0x0020,
	kRtcpNack = 0x0040,
	kRtcpFir = 0x0080,
	kRtcpTmmbr = 0x0100,
	kRtcpTmmbn = 0x0200,
	kRtcpSrReq = 0x0400,
	kRtcpXrVoipMetric = 0x0800,
	kRtcpApp = 0x1000,
	kRtcpRemb = 0x10000,
	kRtcpTransmissionTimeOffset = 0x20000,
	kRtcpXrReceiverReferenceTime = 0x40000,
	kRtcpXrDlrrReportBlock = 0x80000,
	kRtcpTransportFeedback = 0x100000,
	kRtcpXrTargetBitrate = 0x200000
	};

	enum KeyFrameRequestMethod { kKeyFrameReqPliRtcp, kKeyFrameReqFirRtcp };

	enum RtpRtcpPacketType { kPacketRtp = 0, kPacketKeepAlive = 1 };

	// kConditionallyRetransmitHigherLayers allows retransmission of video frames
	// in higher layers if either the last frame in that layer was too far back in
	// time, or if we estimate that a new frame will be available in a lower layer
	// in a shorter time than it would take to request and receive a retransmission.
	enum RetransmissionMode : uint8_t {
	kRetransmitOff = 0x0,
	kRetransmitFECPackets = 0x1,
	kRetransmitBaseLayer = 0x2,
	kRetransmitHigherLayers = 0x4,
	kConditionallyRetransmitHigherLayers = 0x8,
	kRetransmitAllPackets = 0xFF
	};

	enum RtxMode {
	kRtxOff = 0x0,
	kRtxRetransmitted = 0x1, // Only send retransmissions over RTX.
	kRtxRedundantPayloads = 0x2 // Preventively send redundant payloads
	// instead of padding.
	};

	const size_t kRtxHeaderSize = 2;

	struct RTCPReportBlock {
	RTCPReportBlock()
	: sender_ssrc(0),
	source_ssrc(0),
	fraction_lost(0),
	packets_lost(0),
	extended_highest_sequence_number(0),
	jitter(0),
	last_sender_report_timestamp(0),
	delay_since_last_sender_report(0) {}

	RTCPReportBlock(uint32_t sender_ssrc,
	uint32_t source_ssrc,
	uint8_t fraction_lost,
	int32_t packets_lost,
	uint32_t extended_highest_sequence_number,
	uint32_t jitter,
	uint32_t last_sender_report_timestamp,
	uint32_t delay_since_last_sender_report)
	: sender_ssrc(sender_ssrc),
	source_ssrc(source_ssrc),
	fraction_lost(fraction_lost),
	packets_lost(packets_lost),
	extended_highest_sequence_number(extended_highest_sequence_number),
	jitter(jitter),
	last_sender_report_timestamp(last_sender_report_timestamp),
	delay_since_last_sender_report(delay_since_last_sender_report) {}

	// Fields as described by RFC 3550 6.4.2.
	uint32_t sender_ssrc; // SSRC of sender of this report.
	uint32_t source_ssrc; // SSRC of the RTP packet sender.
	uint8_t fraction_lost;
	uint32_t packets_lost; // 24 bits valid.
	uint32_t extended_highest_sequence_number;
	uint32_t jitter;
	uint32_t last_sender_report_timestamp;
	uint32_t delay_since_last_sender_report;
	};

	typedef std::list<RTCPReportBlock> ReportBlockList;

	struct RtpState {
	RtpState()
	: sequence_number(0),
	start_timestamp(0),
	timestamp(0),
	capture_time_ms(-1),
	last_timestamp_time_ms(-1),
	media_has_been_sent(false) {}
	uint16_t sequence_number;
	uint32_t start_timestamp;
	uint32_t timestamp;
	int64_t capture_time_ms;
	int64_t last_timestamp_time_ms;
	bool media_has_been_sent;
	};

	class RtpData {
	public:
	virtual ~RtpData() {}

	virtual int32_t OnReceivedPayloadData(const uint8_t* payload_data,
	size_t payload_size,
	const WebRtcRTPHeader* rtp_header) = 0;
	};

	// Callback interface for packets recovered by FlexFEC or ULPFEC. In
	// the FlexFEC case, the implementation should be able to demultiplex
	// the recovered RTP packets based on SSRC.
	class RecoveredPacketReceiver {
	public:
	virtual void OnRecoveredPacket(const uint8_t* packet, size_t length) = 0;

	protected:
	virtual ~RecoveredPacketReceiver() = default;
	};

	class RtcpIntraFrameObserver {
	public:
	virtual ~RtcpIntraFrameObserver() {}

	virtual void OnReceivedIntraFrameRequest(uint32_t ssrc) = 0;
	};

	class RtcpBandwidthObserver {
	public:
	// REMB or TMMBR
	virtual void OnReceivedEstimatedBitrate(uint32_t bitrate) = 0;

	virtual void OnReceivedRtcpReceiverReport(
	const ReportBlockList& report_blocks,
	int64_t rtt,
	int64_t now_ms) = 0;

	virtual ~RtcpBandwidthObserver() {}
	};

	struct PacketFeedback {
	PacketFeedback(int64_t arrival_time_ms, uint16_t sequence_number)
	: PacketFeedback(-1,
	arrival_time_ms,
	kNoSendTime,
	sequence_number,
	0,
	0,
	0,
	PacedPacketInfo()) {}

	PacketFeedback(int64_t arrival_time_ms,
	int64_t send_time_ms,
	uint16_t sequence_number,
	size_t payload_size,
	const PacedPacketInfo& pacing_info)
	: PacketFeedback(-1,
	arrival_time_ms,
	send_time_ms,
	sequence_number,
	payload_size,
	0,
	0,
	pacing_info) {}

	PacketFeedback(int64_t creation_time_ms,
	uint16_t sequence_number,
	size_t payload_size,
	uint16_t local_net_id,
	uint16_t remote_net_id,
	const PacedPacketInfo& pacing_info)
	: PacketFeedback(creation_time_ms,
	kNotReceived,
	kNoSendTime,
	sequence_number,
	payload_size,
	local_net_id,
	remote_net_id,
	pacing_info) {}

	PacketFeedback(int64_t creation_time_ms,
	int64_t arrival_time_ms,
	int64_t send_time_ms,
	uint16_t sequence_number,
	size_t payload_size,
	uint16_t local_net_id,
	uint16_t remote_net_id,
	const PacedPacketInfo& pacing_info)
	: creation_time_ms(creation_time_ms),
	arrival_time_ms(arrival_time_ms),
	send_time_ms(send_time_ms),
	sequence_number(sequence_number),
	payload_size(payload_size),
	local_net_id(local_net_id),
	remote_net_id(remote_net_id),
	pacing_info(pacing_info) {}

	static constexpr int kNotAProbe = -1;
	static constexpr int64_t kNotReceived = -1;
	static constexpr int64_t kNoSendTime = -1;

	// NOTE! The variable \|creation_time_ms\| is not used when testing equality.
	// This is due to \|creation_time_ms\| only being used by SendTimeHistory
	// for book-keeping, and is of no interest outside that class.
	// TODO(philipel): Remove \|creation_time_ms\| from PacketFeedback when cleaning
	// up SendTimeHistory.
	bool operator==(const PacketFeedback& rhs) const {
	return arrival_time_ms == rhs.arrival_time_ms &&
	send_time_ms == rhs.send_time_ms &&
	sequence_number == rhs.sequence_number &&
	payload_size == rhs.payload_size && pacing_info == rhs.pacing_info;
	}

	// Time corresponding to when this object was created.
	int64_t creation_time_ms;
	// Time corresponding to when the packet was received. Timestamped with the
	// receiver's clock. For unreceived packet, the sentinel value kNotReceived
	// is used.
	int64_t arrival_time_ms;
	// Time corresponding to when the packet was sent, timestamped with the
	// sender's clock.
	int64_t send_time_ms;
	// Packet identifier, incremented with 1 for every packet generated by the
	// sender.
	uint16_t sequence_number;
	// Session unique packet identifier, incremented with 1 for every packet
	// generated by the sender.
	int64_t long_sequence_number;
	// Size of the packet excluding RTP headers.
	size_t payload_size;
	// The network route ids that this packet is associated with.
	uint16_t local_net_id;
	uint16_t remote_net_id;
	// Pacing information about this packet.
	PacedPacketInfo pacing_info;
	};

	class PacketFeedbackComparator {
	public:
	inline bool operator()(const PacketFeedback& lhs, const PacketFeedback& rhs) {
	if (lhs.arrival_time_ms != rhs.arrival_time_ms)
	return lhs.arrival_time_ms < rhs.arrival_time_ms;
	if (lhs.send_time_ms != rhs.send_time_ms)
	return lhs.send_time_ms < rhs.send_time_ms;
	return lhs.sequence_number < rhs.sequence_number;
	}
	};

	class TransportFeedbackObserver {
	public:
	TransportFeedbackObserver() {}
	virtual ~TransportFeedbackObserver() {}

	// Note: Transport-wide sequence number as sequence number.
	virtual void AddPacket(uint32_t ssrc,
	uint16_t sequence_number,
	size_t length,
	const PacedPacketInfo& pacing_info) = 0;

	virtual void OnTransportFeedback(const rtcp::TransportFeedback& feedback) = 0;
	};

	// Interface for PacketRouter to send rtcp feedback on behalf of
	// congestion controller.
	// TODO(bugs.webrtc.org/8239): Remove and use RtcpTransceiver directly
	// when RtcpTransceiver always present in rtp transport.
	class RtcpFeedbackSenderInterface {
	public:
	virtual ~RtcpFeedbackSenderInterface() = default;
	virtual uint32_t SSRC() const = 0;
	virtual bool SendFeedbackPacket(const rtcp::TransportFeedback& feedback) = 0;
	virtual void SetRemb(int64_t bitrate_bps, std::vector<uint32_t> ssrcs) = 0;
	virtual void UnsetRemb() = 0;
	};

	class PacketFeedbackObserver {
	public:
	virtual ~PacketFeedbackObserver() = default;

	virtual void OnPacketAdded(uint32_t ssrc, uint16_t seq_num) = 0;
	virtual void OnPacketFeedbackVector(
	const std::vector<PacketFeedback>& packet_feedback_vector) = 0;
	};

	class RtcpRttStats {
	public:
	virtual void OnRttUpdate(int64_t rtt) = 0;

	virtual int64_t LastProcessedRtt() const = 0;

	virtual ~RtcpRttStats() {}
	};

	// Statistics about packet loss for a single directional connection. All values
	// are totals since the connection initiated.
	struct RtpPacketLossStats {
	// The number of packets lost in events where no adjacent packets were also
	// lost.
	uint64_t single_packet_loss_count;
	// The number of events in which more than one adjacent packet was lost.
	uint64_t multiple_packet_loss_event_count;
	// The number of packets lost in events where more than one adjacent packet
	// was lost.
	uint64_t multiple_packet_loss_packet_count;
	};

	class RtpPacketSender {
	public:
	RtpPacketSender() {}
	virtual ~RtpPacketSender() {}

	enum Priority {
	kHighPriority = 0, // Pass through; will be sent immediately.
	kNormalPriority = 2, // Put in back of the line.
	kLowPriority = 3, // Put in back of the low priority line.
	};
	// Low priority packets are mixed with the normal priority packets
	// while we are paused.

	// Returns true if we send the packet now, else it will add the packet
	// information to the queue and call TimeToSendPacket when it's time to send.
	virtual void InsertPacket(Priority priority,
	uint32_t ssrc,
	uint16_t sequence_number,
	int64_t capture_time_ms,
	size_t bytes,
	bool retransmission) = 0;

	// Currently audio traffic is not accounted by pacer and passed through.
	// With the introduction of audio BWE audio traffic will be accounted for
	// the pacer budget calculation. The audio traffic still will be injected
	// at high priority.
	// TODO(alexnarest): Make it pure virtual after rtp_sender_unittest will be
	// updated to support it
	virtual void SetAccountForAudioPackets(bool account_for_audio) {}
	};

	class TransportSequenceNumberAllocator {
	public:
	TransportSequenceNumberAllocator() {}
	virtual ~TransportSequenceNumberAllocator() {}

	virtual uint16_t AllocateSequenceNumber() = 0;
	};

	struct RtpPacketCounter {
	RtpPacketCounter()
	: header_bytes(0), payload_bytes(0), padding_bytes(0), packets(0) {}

	void Add(const RtpPacketCounter& other) {
	header_bytes += other.header_bytes;
	payload_bytes += other.payload_bytes;
	padding_bytes += other.padding_bytes;
	packets += other.packets;
	}

	void Subtract(const RtpPacketCounter& other) {
	RTC_DCHECK_GE(header_bytes, other.header_bytes);
	header_bytes -= other.header_bytes;
	RTC_DCHECK_GE(payload_bytes, other.payload_bytes);
	payload_bytes -= other.payload_bytes;
	RTC_DCHECK_GE(padding_bytes, other.padding_bytes);
	padding_bytes -= other.padding_bytes;
	RTC_DCHECK_GE(packets, other.packets);
	packets -= other.packets;
	}

	void AddPacket(size_t packet_length, const RTPHeader& header) {
	++packets;
	header_bytes += header.headerLength;
	padding_bytes += header.paddingLength;
	payload_bytes +=
	packet_length - (header.headerLength + header.paddingLength);
	}

	size_t TotalBytes() const {
	return header_bytes + payload_bytes + padding_bytes;
	}

	size_t header_bytes; // Number of bytes used by RTP headers.
	size_t payload_bytes; // Payload bytes, excluding RTP headers and padding.
	size_t padding_bytes; // Number of padding bytes.
	uint32_t packets; // Number of packets.
	};

	// Data usage statistics for a (rtp) stream.
	struct StreamDataCounters {
	StreamDataCounters();

	void Add(const StreamDataCounters& other) {
	transmitted.Add(other.transmitted);
	retransmitted.Add(other.retransmitted);
	fec.Add(other.fec);
	if (other.first_packet_time_ms != -1 &&
	(other.first_packet_time_ms < first_packet_time_ms \|\|
	first_packet_time_ms == -1)) {
	// Use oldest time.
	first_packet_time_ms = other.first_packet_time_ms;
	}
	}

	void Subtract(const StreamDataCounters& other) {
	transmitted.Subtract(other.transmitted);
	retransmitted.Subtract(other.retransmitted);
	fec.Subtract(other.fec);
	if (other.first_packet_time_ms != -1 &&
	(other.first_packet_time_ms > first_packet_time_ms \|\|
	first_packet_time_ms == -1)) {
	// Use youngest time.
	first_packet_time_ms = other.first_packet_time_ms;
	}
	}

	int64_t TimeSinceFirstPacketInMs(int64_t now_ms) const {
	return (first_packet_time_ms == -1) ? -1 : (now_ms - first_packet_time_ms);
	}

	// Returns the number of bytes corresponding to the actual media payload (i.e.
	// RTP headers, padding, retransmissions and fec packets are excluded).
	// Note this function does not have meaning for an RTX stream.
	size_t MediaPayloadBytes() const {
	return transmitted.payload_bytes - retransmitted.payload_bytes -
	fec.payload_bytes;
	}

	int64_t first_packet_time_ms; // Time when first packet is sent/received.
	RtpPacketCounter transmitted; // Number of transmitted packets/bytes.
	RtpPacketCounter retransmitted; // Number of retransmitted packets/bytes.
	RtpPacketCounter fec; // Number of redundancy packets/bytes.
	};

	// Callback, called whenever byte/packet counts have been updated.
	class StreamDataCountersCallback {
	public:
	virtual ~StreamDataCountersCallback() {}

	virtual void DataCountersUpdated(const StreamDataCounters& counters,
	uint32_t ssrc) = 0;
	};

	} // namespace webrtc
	#endif // MODULES_RTP_RTCP_INCLUDE_RTP_RTCP_DEFINES_H_