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Diffstat (limited to 'webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h')
| -rw-r--r-- | webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h | 440 |
1 files changed, 0 insertions, 440 deletions
diff --git a/webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h b/webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h deleted file mode 100644 index 6936352aca..0000000000 --- a/webrtc/modules/rtp_rtcp/interface/rtp_rtcp_defines.h +++ /dev/null @@ -1,440 +0,0 @@ -/* - * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved. - * - * Use of this source code is governed by a BSD-style license - * that can be found in the LICENSE file in the root of the source - * tree. An additional intellectual property rights grant can be found - * in the file PATENTS. All contributing project authors may - * be found in the AUTHORS file in the root of the source tree. - */ - -#ifndef WEBRTC_MODULES_RTP_RTCP_INTERFACE_RTP_RTCP_DEFINES_H_ -#define WEBRTC_MODULES_RTP_RTCP_INTERFACE_RTP_RTCP_DEFINES_H_ - -#include <stddef.h> -#include <list> - -#include "webrtc/modules/interface/module_common_types.h" -#include "webrtc/system_wrappers/include/clock.h" -#include "webrtc/typedefs.h" - -#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 -#define TIMEOUT_SEI_MESSAGES_MS 30000 // in milliseconds - -namespace webrtc { -namespace rtcp { -class TransportFeedback; -} - -const int kVideoPayloadTypeFrequency = 90000; - -// Minimum RTP header size in bytes. -const uint8_t kRtpHeaderSize = 12; - -struct AudioPayload -{ - uint32_t frequency; - uint8_t channels; - uint32_t rate; -}; - -struct VideoPayload -{ - RtpVideoCodecTypes videoCodecType; - uint32_t maxRate; -}; - -union PayloadUnion -{ - AudioPayload Audio; - VideoPayload Video; -}; - -enum RTPAliveType -{ - kRtpDead = 0, - kRtpNoRtp = 1, - kRtpAlive = 2 -}; - -enum ProtectionType { - kUnprotectedPacket, - kProtectedPacket -}; - -enum StorageType { - kDontRetransmit, - kAllowRetransmission -}; - -enum RTPExtensionType { - kRtpExtensionNone, - kRtpExtensionTransmissionTimeOffset, - kRtpExtensionAudioLevel, - kRtpExtensionAbsoluteSendTime, - kRtpExtensionVideoRotation, - kRtpExtensionTransportSequenceNumber, -}; - -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, - kRtcpSli = 0x4000, - kRtcpRpsi = 0x8000, - kRtcpRemb = 0x10000, - kRtcpTransmissionTimeOffset = 0x20000, - kRtcpXrReceiverReferenceTime = 0x40000, - kRtcpXrDlrrReportBlock = 0x80000, - kRtcpTransportFeedback = 0x100000, -}; - -enum KeyFrameRequestMethod { kKeyFrameReqPliRtcp, kKeyFrameReqFirRtcp }; - -enum RtpRtcpPacketType -{ - kPacketRtp = 0, - kPacketKeepAlive = 1 -}; - -enum NACKMethod -{ - kNackOff = 0, - kNackRtcp = 2 -}; - -enum RetransmissionMode : uint8_t { - kRetransmitOff = 0x0, - kRetransmitFECPackets = 0x1, - kRetransmitBaseLayer = 0x2, - kRetransmitHigherLayers = 0x4, - 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 RTCPSenderInfo -{ - uint32_t NTPseconds; - uint32_t NTPfraction; - uint32_t RTPtimeStamp; - uint32_t sendPacketCount; - uint32_t sendOctetCount; -}; - -struct RTCPReportBlock { - RTCPReportBlock() - : remoteSSRC(0), sourceSSRC(0), fractionLost(0), cumulativeLost(0), - extendedHighSeqNum(0), jitter(0), lastSR(0), - delaySinceLastSR(0) {} - - RTCPReportBlock(uint32_t remote_ssrc, - uint32_t source_ssrc, - uint8_t fraction_lost, - uint32_t cumulative_lost, - uint32_t extended_high_sequence_number, - uint32_t jitter, - uint32_t last_sender_report, - uint32_t delay_since_last_sender_report) - : remoteSSRC(remote_ssrc), - sourceSSRC(source_ssrc), - fractionLost(fraction_lost), - cumulativeLost(cumulative_lost), - extendedHighSeqNum(extended_high_sequence_number), - jitter(jitter), - lastSR(last_sender_report), - delaySinceLastSR(delay_since_last_sender_report) {} - - // Fields as described by RFC 3550 6.4.2. - uint32_t remoteSSRC; // SSRC of sender of this report. - uint32_t sourceSSRC; // SSRC of the RTP packet sender. - uint8_t fractionLost; - uint32_t cumulativeLost; // 24 bits valid. - uint32_t extendedHighSeqNum; - uint32_t jitter; - uint32_t lastSR; - uint32_t delaySinceLastSR; -}; - -struct RtcpReceiveTimeInfo { - // Fields as described by RFC 3611 4.5. - uint32_t sourceSSRC; - uint32_t lastRR; - uint32_t delaySinceLastRR; -}; - -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* payloadData, - const size_t payloadSize, - const WebRtcRTPHeader* rtpHeader) = 0; - - virtual bool OnRecoveredPacket(const uint8_t* packet, - size_t packet_length) = 0; -}; - -class RtpFeedback -{ -public: - virtual ~RtpFeedback() {} - - // Receiving payload change or SSRC change. (return success!) - /* - * channels - number of channels in codec (1 = mono, 2 = stereo) - */ - virtual int32_t OnInitializeDecoder( - const int8_t payloadType, - const char payloadName[RTP_PAYLOAD_NAME_SIZE], - const int frequency, - const uint8_t channels, - const uint32_t rate) = 0; - - virtual void OnIncomingSSRCChanged(const uint32_t ssrc) = 0; - - virtual void OnIncomingCSRCChanged(const uint32_t CSRC, - const bool added) = 0; -}; - -class RtpAudioFeedback { - public: - virtual void OnPlayTelephoneEvent(const uint8_t event, - const uint16_t lengthMs, - const uint8_t volume) = 0; - - protected: - virtual ~RtpAudioFeedback() {} -}; - -class RtcpIntraFrameObserver { - public: - virtual void OnReceivedIntraFrameRequest(uint32_t ssrc) = 0; - - virtual void OnReceivedSLI(uint32_t ssrc, - uint8_t picture_id) = 0; - - virtual void OnReceivedRPSI(uint32_t ssrc, - uint64_t picture_id) = 0; - - virtual void OnLocalSsrcChanged(uint32_t old_ssrc, uint32_t new_ssrc) = 0; - - virtual ~RtcpIntraFrameObserver() {} -}; - -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 PacketInfo { - PacketInfo(int64_t arrival_time_ms, uint16_t sequence_number) - : PacketInfo(-1, arrival_time_ms, -1, sequence_number, 0, false) {} - - PacketInfo(int64_t arrival_time_ms, - int64_t send_time_ms, - uint16_t sequence_number, - size_t payload_size, - bool was_paced) - : PacketInfo(-1, - arrival_time_ms, - send_time_ms, - sequence_number, - payload_size, - was_paced) {} - - PacketInfo(int64_t creation_time_ms, - int64_t arrival_time_ms, - int64_t send_time_ms, - uint16_t sequence_number, - size_t payload_size, - bool was_paced) - : 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), - was_paced(was_paced) {} - - // 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. - 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; - // Size of the packet excluding RTP headers. - size_t payload_size; - // True if the packet was paced out by the pacer. - bool was_paced; -}; - -class TransportFeedbackObserver { - public: - TransportFeedbackObserver() {} - virtual ~TransportFeedbackObserver() {} - - // Note: Transport-wide sequence number as sequence number. Arrival time - // must be set to 0. - virtual void AddPacket(uint16_t sequence_number, - size_t length, - bool was_paced) = 0; - - virtual void OnTransportFeedback(const rtcp::TransportFeedback& feedback) = 0; -}; - -class RtcpRttStats { - public: - virtual void OnRttUpdate(int64_t rtt) = 0; - - virtual int64_t LastProcessedRtt() const = 0; - - virtual ~RtcpRttStats() {}; -}; - -// Null object version of RtpFeedback. -class NullRtpFeedback : public RtpFeedback { - public: - virtual ~NullRtpFeedback() {} - - int32_t OnInitializeDecoder(const int8_t payloadType, - const char payloadName[RTP_PAYLOAD_NAME_SIZE], - const int frequency, - const uint8_t channels, - const uint32_t rate) override { - return 0; - } - - void OnIncomingSSRCChanged(const uint32_t ssrc) override {} - void OnIncomingCSRCChanged(const uint32_t CSRC, const bool added) override {} -}; - -// Null object version of RtpData. -class NullRtpData : public RtpData { - public: - virtual ~NullRtpData() {} - - int32_t OnReceivedPayloadData(const uint8_t* payloadData, - const size_t payloadSize, - const WebRtcRTPHeader* rtpHeader) override { - return 0; - } - - bool OnRecoveredPacket(const uint8_t* packet, size_t packet_length) override { - return true; - } -}; - -// Null object version of RtpAudioFeedback. -class NullRtpAudioFeedback : public RtpAudioFeedback { - public: - virtual ~NullRtpAudioFeedback() {} - - void OnPlayTelephoneEvent(const uint8_t event, - const uint16_t lengthMs, - const uint8_t volume) override {} -}; - -// 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; -}; - -class TransportSequenceNumberAllocator { - public: - TransportSequenceNumberAllocator() {} - virtual ~TransportSequenceNumberAllocator() {} - - virtual uint16_t AllocateSequenceNumber() = 0; -}; - -} // namespace webrtc -#endif // WEBRTC_MODULES_RTP_RTCP_INTERFACE_RTP_RTCP_DEFINES_H_ |