Rtp.h

Go to the documentation of this file.

#ifndef __KINESIS_VIDEO_WEBRTC_CLIENT_PEERCONNECTION_RTP__
#define __KINESIS_VIDEO_WEBRTC_CLIENT_PEERCONNECTION_RTP__
 
#pragma once
 
#ifdef __cplusplus
extern "C" {
#endif
 
// Default MTU comes from libwebrtc
// https://groups.google.com/forum/#!topic/discuss-webrtc/gH5ysR3SoZI
#define DEFAULT_MTU_SIZE_BYTES                     1200
#define DEFAULT_ROLLING_BUFFER_DURATION_IN_SECONDS 3
#define DEFAULT_EXPECTED_VIDEO_BIT_RATE            (5 * 1024 * 1024)
// Opus has highest based on the supported codecs which is 510Kibps. So setting this to twice
#define DEFAULT_EXPECTED_AUDIO_BIT_RATE        (1000 * 1024)
#define DEFAULT_SEQ_NUM_BUFFER_SIZE            1000
#define DEFAULT_VALID_INDEX_BUFFER_SIZE        1000
#define DEFAULT_PEER_FRAME_BUFFER_SIZE         (5 * 1024)
#define SRTP_AUTH_TAG_OVERHEAD                 10
#define MIN_ROLLING_BUFFER_DURATION_IN_SECONDS (DOUBLE) 0.1
#define MIN_EXPECTED_BIT_RATE                  (DOUBLE)(102.4 * 1024) // Considering 1Kib = 1024 bits
 
// https://www.w3.org/TR/webrtc-stats/#dom-rtcoutboundrtpstreamstats-huge
// Huge frames, by definition, are frames that have an encoded size at least 2.5 times the average size of the frames.
#define HUGE_FRAME_MULTIPLIER 2.5
 
typedef struct {
    UINT8 payloadType;
    UINT8 rtxPayloadType;
    UINT16 sequenceNumber;
    UINT16 rtxSequenceNumber;
    UINT32 ssrc;
    UINT32 rtxSsrc;
    PayloadArray payloadArray;
 
    RtcMediaStreamTrack track;
    PRtpRollingBuffer packetBuffer;
    PRetransmitter retransmitter;
 
    UINT64 rtpTimeOffset;
    UINT64 firstFrameWallClockTime; // 100ns precision
 
    // used for fps calculation
    UINT64 lastKnownFrameCount;
    UINT64 lastKnownFrameCountTime; // 100ns precision
 
} RtcRtpSender, *PRtcRtpSender;
 
typedef struct {
    RtcRtpTransceiver transceiver;
    RtcRtpSender sender;
 
    PKvsPeerConnection pKvsPeerConnection;
 
    DOUBLE rollingBufferDurationSec;
    DOUBLE rollingBufferBitratebps;
 
    UINT32 jitterBufferSsrc;
    PJitterBuffer pJitterBuffer;
 
    UINT64 onFrameCustomData;
    RtcOnFrame onFrame;
 
    UINT64 onBandwidthEstimationCustomData;
    RtcOnBandwidthEstimation onBandwidthEstimation;
    UINT64 onPictureLossCustomData;
    RtcOnPictureLoss onPictureLoss;
 
    PBYTE peerFrameBuffer;
    UINT32 peerFrameBufferSize;
 
    UINT32 rtcpReportsTimerId;
 
    MUTEX statsLock;
    RtcOutboundRtpStreamStats outboundStats;
    RtcRemoteInboundRtpStreamStats remoteInboundStats;
    RtcInboundRtpStreamStats inboundStats;
} KvsRtpTransceiver, *PKvsRtpTransceiver;
 
STATUS createKvsRtpTransceiver(RTC_RTP_TRANSCEIVER_DIRECTION, DOUBLE, DOUBLE, PKvsPeerConnection, UINT32, UINT32, PRtcMediaStreamTrack, PJitterBuffer,
                               RTC_CODEC, PKvsRtpTransceiver*);
STATUS freeKvsRtpTransceiver(PKvsRtpTransceiver*);
 
STATUS kvsRtpTransceiverSetJitterBuffer(PKvsRtpTransceiver, PJitterBuffer);
 
#define CONVERT_TIMESTAMP_TO_RTP(clockRate, pts) ((UINT64) ((DOUBLE) (pts) * ((DOUBLE) (clockRate) / HUNDREDS_OF_NANOS_IN_A_SECOND)))
 
STATUS writeRtpPacket(PKvsPeerConnection pKvsPeerConnection, PRtpPacket pRtpPacket);
 
STATUS hasTransceiverWithSsrc(PKvsPeerConnection pKvsPeerConnection, UINT32 ssrc);
STATUS findTransceiverBySsrc(PKvsPeerConnection pKvsPeerConnection, PKvsRtpTransceiver* ppTransceiver, UINT32 ssrc);
 
#ifdef __cplusplus
}
#endif
#endif /* __KINESIS_VIDEO_WEBRTC_CLIENT_PEERCONNECTION_RTP__ */