RtcOutboundRtpStreamStats Gathers stats for media stream from the embedded device Note: RTCOutboundRtpStreamStats extends RTCSentRtpStreamStats as per https://www.w3.org/TR/webrtc-stats/#dom-rtcoutboundrtpstreamstats. More...

#include <Stats.h>

Collaboration diagram for RtcOutboundRtpStreamStats:

Public Attributes
RTCSentRtpStreamStats	sent
	Comprises of information such as packetsSent and bytesSent. More...

BOOL	voiceActivityFlag

DOMString	trackId
	ID representing current track attached to the sender of the stream. More...

DOMString	mediaSourceId
	TODO ID representing the current media source. More...

DOMString	senderId
	TODO The stats ID used to look up the RTCAudioSenderStats or RTCVideoSenderStats object sending this stream. More...

DOMString	remoteId
	TODO ID to look up the remote RTCRemoteInboundRtpStreamStats object for the same SSRC. More...

DOMString	rid
	TODO Exposes the rid encoding parameter of this RTP stream if it has been set, otherwise it is undefined. More...

DOMString	encoderImplementation
	Identifies the encoder implementation used. More...

UINT32	packetsDiscardedOnSend
	Total number of RTP packets for this SSRC that have been discarded due to socket errors. More...

UINT32	framesSent
	Only valid for video. Represents the total number of frames sent on this RTP stream. More...

UINT32	hugeFramesSent

UINT32	framesEncoded
	Only valid for video. It represents the total number of frames successfully encoded for this RTP media stream. More...

UINT32	keyFramesEncoded
	Only valid for video. It represents the total number of key frames encoded successfully in the RTP Stream. More...

UINT32	framesDiscardedOnSend
	Total number of video frames that have been discarded for this SSRC due to socket errors. More...

UINT32	frameWidth
	Only valid for video. Represents the width of the last encoded frame. More...

UINT32	frameHeight
	Only valid for video. Represents the height of the last encoded frame. More...

UINT32	frameBitDepth
	Only valid for video. Represents the bit depth per pixel of the last encoded frame. Typical values: 24, 30, 36. More...

UINT32	nackCount
	Count the total number of Negative ACKnowledgement (NACK) packets received by this sender. More...

UINT32	firCount
	Only valid for video. Count the total number of Full Intra Request (FIR) packets received by this sender. More...

UINT32	pliCount
	Only valid for video. Count the total number of Picture Loss Indication (PLI) packets received by this sender. More...

UINT32	sliCount
	Only valid for video. Count the total number of Slice Loss Indication (SLI) packets received by this sender. More...

UINT32	qualityLimitationResolutionChanges

INT32	fecPacketsSent
	TODO Total number of RTP FEC packets sent for this SSRC. Can also be incremented while sending FEC packets in band. More...

UINT64	lastPacketSentTimestamp
	The timestamp in milliseconds at which the last packet was sent for this SSRC. More...

UINT64	headerBytesSent
	Total number of RTP header and padding bytes sent for this SSRC. More...

UINT64	bytesDiscardedOnSend
	Total number of bytes for this SSRC that have been discarded due to socket errors. More...

UINT64	retransmittedPacketsSent
	The total number of packets that were retransmitted for this SSRC. More...

UINT64	retransmittedBytesSent
	The total number of PAYLOAD bytes retransmitted for this SSRC. More...

UINT64	targetBitrate
	Current target TIAS bitrate configured for this particular SSRC. More...

UINT64	totalEncodedBytesTarget
	Increased by the target frame size in bytes every time a frame has been encoded. More...

DOUBLE	framesPerSecond
	Only valid for video. The number of encoded frames during the last second. More...

UINT64	qpSum
	TODO Only valid for video. The sum of the QP values of frames encoded by this sender. QP value depends on the codec. More...

UINT64	totalSamplesSent
	TODO Only valid for audio. The total number of samples that have been sent over this RTP stream. More...

UINT64	samplesEncodedWithSilk
	TODO Only valid for audio and when the audio codec is Opus. Represnets only SILK portion of codec. More...

UINT64	samplesEncodedWithCelt
	TODO Only valid for audio and when the audio codec is Opus. Represnets only CELT portion of codec. More...

UINT64	totalEncodeTime
	Total number of milliseconds that has been spent encoding the framesEncoded frames of the stream. More...

UINT64	totalPacketSendDelay
	Total time (seconds) packets have spent buffered locally before being transmitted onto the network. More...

UINT64	averageRtcpInterval
	The average RTCP interval between two consecutive compound RTCP packets. More...

QualityLimitationDurationsRecord	qualityLimitationDurations
	Total time (seconds) spent in each reason state. More...

DscpPacketsSentRecord	perDscpPacketsSent
	Total number of packets sent for this SSRC, per DSCP. More...

RTC_QUALITY_LIMITATION_REASON	qualityLimitationReason
	Only valid for video. More...

Detailed Description

RtcOutboundRtpStreamStats Gathers stats for media stream from the embedded device Note: RTCOutboundRtpStreamStats extends RTCSentRtpStreamStats as per https://www.w3.org/TR/webrtc-stats/#dom-rtcoutboundrtpstreamstats.

Reference: https://www.w3.org/TR/webrtc-stats/#outboundrtpstats-dict*

Member Data Documentation

◆ averageRtcpInterval

UINT64 RtcOutboundRtpStreamStats::averageRtcpInterval

The average RTCP interval between two consecutive compound RTCP packets.

◆ bytesDiscardedOnSend

UINT64 RtcOutboundRtpStreamStats::bytesDiscardedOnSend

Total number of bytes for this SSRC that have been discarded due to socket errors.

◆ encoderImplementation

DOMString RtcOutboundRtpStreamStats::encoderImplementation

Identifies the encoder implementation used.

◆ fecPacketsSent

INT32 RtcOutboundRtpStreamStats::fecPacketsSent

TODO Total number of RTP FEC packets sent for this SSRC. Can also be incremented while sending FEC packets in band.

◆ firCount

UINT32 RtcOutboundRtpStreamStats::firCount

Only valid for video. Count the total number of Full Intra Request (FIR) packets received by this sender.

◆ frameBitDepth

UINT32 RtcOutboundRtpStreamStats::frameBitDepth

Only valid for video. Represents the bit depth per pixel of the last encoded frame. Typical values: 24, 30, 36.

◆ frameHeight

UINT32 RtcOutboundRtpStreamStats::frameHeight

Only valid for video. Represents the height of the last encoded frame.

◆ framesDiscardedOnSend

UINT32 RtcOutboundRtpStreamStats::framesDiscardedOnSend

Total number of video frames that have been discarded for this SSRC due to socket errors.

◆ framesEncoded

UINT32 RtcOutboundRtpStreamStats::framesEncoded

Only valid for video. It represents the total number of frames successfully encoded for this RTP media stream.

◆ framesPerSecond

DOUBLE RtcOutboundRtpStreamStats::framesPerSecond

Only valid for video. The number of encoded frames during the last second.

◆ framesSent

UINT32 RtcOutboundRtpStreamStats::framesSent

Only valid for video. Represents the total number of frames sent on this RTP stream.

◆ frameWidth

UINT32 RtcOutboundRtpStreamStats::frameWidth

Only valid for video. Represents the width of the last encoded frame.

◆ headerBytesSent

UINT64 RtcOutboundRtpStreamStats::headerBytesSent

Total number of RTP header and padding bytes sent for this SSRC.

◆ hugeFramesSent

UINT32 RtcOutboundRtpStreamStats::hugeFramesSent

Only valid for video. Represents the total number of huge frames sent by this RTP stream Huge frames have an encoded size at least 2.5 times the average size of the frames

◆ keyFramesEncoded

UINT32 RtcOutboundRtpStreamStats::keyFramesEncoded

Only valid for video. It represents the total number of key frames encoded successfully in the RTP Stream.

◆ lastPacketSentTimestamp

UINT64 RtcOutboundRtpStreamStats::lastPacketSentTimestamp

The timestamp in milliseconds at which the last packet was sent for this SSRC.

◆ mediaSourceId

DOMString RtcOutboundRtpStreamStats::mediaSourceId

TODO ID representing the current media source.

◆ nackCount

UINT32 RtcOutboundRtpStreamStats::nackCount

Count the total number of Negative ACKnowledgement (NACK) packets received by this sender.

◆ packetsDiscardedOnSend

UINT32 RtcOutboundRtpStreamStats::packetsDiscardedOnSend

Total number of RTP packets for this SSRC that have been discarded due to socket errors.

◆ perDscpPacketsSent

DscpPacketsSentRecord RtcOutboundRtpStreamStats::perDscpPacketsSent

Total number of packets sent for this SSRC, per DSCP.

◆ pliCount

UINT32 RtcOutboundRtpStreamStats::pliCount

Only valid for video. Count the total number of Picture Loss Indication (PLI) packets received by this sender.

◆ qpSum

UINT64 RtcOutboundRtpStreamStats::qpSum

TODO Only valid for video. The sum of the QP values of frames encoded by this sender. QP value depends on the codec.

◆ qualityLimitationDurations

QualityLimitationDurationsRecord RtcOutboundRtpStreamStats::qualityLimitationDurations

Total time (seconds) spent in each reason state.

◆ qualityLimitationReason

RTC_QUALITY_LIMITATION_REASON RtcOutboundRtpStreamStats::qualityLimitationReason

Only valid for video.

◆ qualityLimitationResolutionChanges

UINT32 RtcOutboundRtpStreamStats::qualityLimitationResolutionChanges

Only valid for video. The number of times that the resolution has changed because we are quality limited

◆ remoteId

DOMString RtcOutboundRtpStreamStats::remoteId

TODO ID to look up the remote RTCRemoteInboundRtpStreamStats object for the same SSRC.

◆ retransmittedBytesSent

UINT64 RtcOutboundRtpStreamStats::retransmittedBytesSent

The total number of PAYLOAD bytes retransmitted for this SSRC.

◆ retransmittedPacketsSent

UINT64 RtcOutboundRtpStreamStats::retransmittedPacketsSent

The total number of packets that were retransmitted for this SSRC.

◆ rid

DOMString RtcOutboundRtpStreamStats::rid

TODO Exposes the rid encoding parameter of this RTP stream if it has been set, otherwise it is undefined.

◆ samplesEncodedWithCelt

UINT64 RtcOutboundRtpStreamStats::samplesEncodedWithCelt

TODO Only valid for audio and when the audio codec is Opus. Represnets only CELT portion of codec.

◆ samplesEncodedWithSilk

UINT64 RtcOutboundRtpStreamStats::samplesEncodedWithSilk

TODO Only valid for audio and when the audio codec is Opus. Represnets only SILK portion of codec.

◆ senderId

DOMString RtcOutboundRtpStreamStats::senderId

TODO The stats ID used to look up the RTCAudioSenderStats or RTCVideoSenderStats object sending this stream.

◆ sent

RTCSentRtpStreamStats RtcOutboundRtpStreamStats::sent

Comprises of information such as packetsSent and bytesSent.

◆ sliCount

UINT32 RtcOutboundRtpStreamStats::sliCount

Only valid for video. Count the total number of Slice Loss Indication (SLI) packets received by this sender.

◆ targetBitrate

UINT64 RtcOutboundRtpStreamStats::targetBitrate

Current target TIAS bitrate configured for this particular SSRC.

◆ totalEncodedBytesTarget

UINT64 RtcOutboundRtpStreamStats::totalEncodedBytesTarget

Increased by the target frame size in bytes every time a frame has been encoded.

◆ totalEncodeTime

UINT64 RtcOutboundRtpStreamStats::totalEncodeTime

Total number of milliseconds that has been spent encoding the framesEncoded frames of the stream.

◆ totalPacketSendDelay

UINT64 RtcOutboundRtpStreamStats::totalPacketSendDelay

Total time (seconds) packets have spent buffered locally before being transmitted onto the network.

◆ totalSamplesSent

UINT64 RtcOutboundRtpStreamStats::totalSamplesSent

TODO Only valid for audio. The total number of samples that have been sent over this RTP stream.

◆ trackId

DOMString RtcOutboundRtpStreamStats::trackId

ID representing current track attached to the sender of the stream.

◆ voiceActivityFlag

BOOL RtcOutboundRtpStreamStats::voiceActivityFlag

Only valid for audio. Whether the last RTP packet sent contained voice activity or not based on the presence of the V bit in the extension header

The documentation for this struct was generated from the following file:

src/include/com/amazonaws/kinesis/video/webrtcclient/Stats.h

Public Attributes