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Amazon Kinesis Webrtc C SDK
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Amazon Kinesis Webrtc C SDK
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The RTCInboundRtpStreamStats dictionary represents the measurement metrics for the incoming RTP media stream. The timestamp reported in the statistics object is the time at which the data was sampled. More...
#include <Stats.h>
Public Attributes | |
| RtcReceivedRtpStreamStats | received |
| DOMString | trackId |
| DOMString | receiverId |
| TODO The stats ID used to look up the RTCAudioReceiverStats or RTCVideoReceiverStats object receiving this stream. More... | |
| DOMString | remoteId |
| TODO The remoteId is used for looking up the remote RTCRemoteOutboundRtpStreamStats object for the same SSRC. More... | |
| UINT32 | framesDecoded |
| UINT32 | keyFramesDecoded |
| UINT16 | frameWidth |
| UINT16 | frameHeight |
| UINT8 | frameBitDepth |
| DOUBLE | framesPerSecond |
| TODO Only valid for video. The number of decoded frames in the last second. More... | |
| UINT64 | qpSum |
| DOUBLE | totalDecodeTime |
| DOUBLE | totalInterFrameDelay |
| DOUBLE | totalSquaredInterFrameDelay |
| BOOL | voiceActivityFlag |
| DOMHighResTimeStamp | lastPacketReceivedTimestamp |
| DOUBLE | averageRtcpInterval |
| UINT64 | headerBytesReceived |
| UINT64 | fecPacketsReceived |
| UINT64 | fecPacketsDiscarded |
| UINT64 | bytesReceived |
| Total number of bytes received for this SSRC. Calculated as defined in [RFC3550] section 6.4.1. More... | |
| UINT64 | packetsFailedDecryption |
| UINT64 | packetsDuplicated |
| UINT32 | nackCount |
| TODO Count the total number of Negative ACKnowledgement (NACK) packets sent by this receiver. More... | |
| UINT32 | firCount |
| TODO Only valid for video. Count the total number of Full Intra Request (FIR) packets sent by this receiver. More... | |
| UINT32 | pliCount |
| TODO Only valid for video. Count the total number of Picture Loss Indication (PLI) packets sent by this receiver. More... | |
| UINT32 | sliCount |
| TODO Only valid for video. Count the total number of Slice Loss Indication (SLI) packets sent by this receiver. More... | |
| DOMHighResTimeStamp | estimatedPlayoutTimestamp |
| TODO This is the estimated playout time of this receiver's track. More... | |
| DOUBLE | jitterBufferDelay |
| UINT64 | jitterBufferEmittedCount |
| UINT64 | totalSamplesReceived |
| UINT64 | samplesDecodedWithSilk |
| UINT64 | samplesDecodedWithCelt |
| UINT64 | concealedSamples |
| UINT64 | silentConcealedSamples |
| TODO Only valid for audio. The total number of concealed samples inserted that are "silent". More... | |
| UINT64 | concealmentEvents |
| UINT64 | insertedSamplesForDeceleration |
| UINT64 | removedSamplesForAcceleration |
| DOUBLE | audioLevel |
| DOUBLE | totalAudioEnergy |
| DOUBLE | totalSamplesDuration |
| UINT32 | framesReceived |
| DOMString | decoderImplementation |
| TODO Identifies the decoder implementation used. This is useful for diagnosing interoperability issues. More... | |
The RTCInboundRtpStreamStats dictionary represents the measurement metrics for the incoming RTP media stream. The timestamp reported in the statistics object is the time at which the data was sampled.
Reference: https://www.w3.org/TR/webrtc-stats/#dom-rtcinboundrtpstreamstats
| DOUBLE RtcInboundRtpStreamStats::audioLevel |
TODO Only valid for audio. Represents the audio level of the receiving track. For audio levels of tracks attached locally, see RTCAudioSourceStats instead. The value is between 0..1 (linear), where 1.0 represents 0 dBov, 0 represents silence, and 0.5 represents approximately 6 dBSPL change in the sound pressure level from 0 dBov.The audioLevel is averaged over some small interval, using the algortihm described under totalAudioEnergy. The interval used is implementation dependent.
| DOUBLE RtcInboundRtpStreamStats::averageRtcpInterval |
TODO The average RTCP interval between two consecutive compound RTCP packets. This is calculated by the sending endpoint when sending compound RTCP reports. Compound packets must contain at least a RTCP RR or SR block and an SDES packet with the CNAME item.
| UINT64 RtcInboundRtpStreamStats::bytesReceived |
Total number of bytes received for this SSRC. Calculated as defined in [RFC3550] section 6.4.1.
| UINT64 RtcInboundRtpStreamStats::concealedSamples |
TODO Only valid for audio. The total number of samples that are concealed samples. A concealed sample is a sample that was replaced with synthesized samples generated locally before being played out.
| UINT64 RtcInboundRtpStreamStats::concealmentEvents |
TODO Only valid for audio. The number of concealment events. This counter increases every time a concealed sample is synthesized after a non-concealed sample.
| DOMString RtcInboundRtpStreamStats::decoderImplementation |
TODO Identifies the decoder implementation used. This is useful for diagnosing interoperability issues.
| DOMHighResTimeStamp RtcInboundRtpStreamStats::estimatedPlayoutTimestamp |
TODO This is the estimated playout time of this receiver's track.
| UINT64 RtcInboundRtpStreamStats::fecPacketsDiscarded |
TODO Total number of RTP FEC packets received for this SSRC where the error correction payload was discarded by the application. This may happen 1. if all the source packets protected by the FEC packet were received or already recovered by a separate FEC packet, or 2. if the FEC packet arrived late, i.e., outside the recovery window, and the lost RTP packets have already been skipped during playout. This is a subset of fecPacketsReceived.
| UINT64 RtcInboundRtpStreamStats::fecPacketsReceived |
TODO Total number of RTP FEC packets received for this SSRC. This counter can also be incremented when receiving FEC packets in-band with media packets (e.g., with Opus).
| UINT32 RtcInboundRtpStreamStats::firCount |
TODO Only valid for video. Count the total number of Full Intra Request (FIR) packets sent by this receiver.
| UINT8 RtcInboundRtpStreamStats::frameBitDepth |
TODO Only valid for video. Represents the bit depth per pixel of the last decoded frame. Typical values are 24, 30, or 36 bits. Before the first frame is decoded this attribute is missing.
| UINT16 RtcInboundRtpStreamStats::frameHeight |
TODO Only valid for video. Represents the height of the last decoded frame. Before the first frame is decoded this attribute is missing.
| UINT32 RtcInboundRtpStreamStats::framesDecoded |
TODO Only valid for video. It represents the total number of frames correctly decoded for this RTP stream, i.e., frames that would be displayed if no frames are dropped.
| DOUBLE RtcInboundRtpStreamStats::framesPerSecond |
TODO Only valid for video. The number of decoded frames in the last second.
| UINT32 RtcInboundRtpStreamStats::framesReceived |
Only valid for video. Represents the total number of complete frames received on this RTP stream. This metric is incremented when the complete frame is received.
| UINT16 RtcInboundRtpStreamStats::frameWidth |
TODO Only valid for video. Represents the width of the last decoded frame. Before the first frame is decoded this attribute is missing.
| UINT64 RtcInboundRtpStreamStats::headerBytesReceived |
Total number of RTP header and padding bytes received for this SSRC. This does not include the size of transport layer headers such as IP or UDP. headerBytesReceived + bytesReceived equals the number of bytes received as payload over the transport.
| UINT64 RtcInboundRtpStreamStats::insertedSamplesForDeceleration |
TODO Only valid for audio. When playout is slowed down, this counter is increased by the difference between the number of samples received and the number of samples played out.
| DOUBLE RtcInboundRtpStreamStats::jitterBufferDelay |
TODO It is the sum of the time, in seconds, each audio sample or video frame takes from the time it is received and to the time it exits the jitter buffer.
| UINT64 RtcInboundRtpStreamStats::jitterBufferEmittedCount |
TODO The total number of audio samples or video frames that have come out of the jitter buffer (increasing jitterBufferDelay).
| UINT32 RtcInboundRtpStreamStats::keyFramesDecoded |
TODO Only valid for video. It represents the total number of key frames, such as key frames in VP8 [RFC6386] or IDR-frames in H.264 [RFC6184], successfully decoded for this RTP media stream. This is a subset of framesDecoded. framesDecoded - keyFramesDecoded gives you the number of delta frames decoded.
| DOMHighResTimeStamp RtcInboundRtpStreamStats::lastPacketReceivedTimestamp |
Represents the timestamp at which the last packet was received for this SSRC. This differs from timestamp, which represents the time at which the statistics were generated by the local endpoint.
| UINT32 RtcInboundRtpStreamStats::nackCount |
TODO Count the total number of Negative ACKnowledgement (NACK) packets sent by this receiver.
| UINT64 RtcInboundRtpStreamStats::packetsDuplicated |
TODO The cumulative number of packets discarded because they are duplicated. Duplicate packets are not counted in packetsDiscarded. Duplicated packets have the same RTP sequence number and content as a previously received packet. If multiple duplicates of a packet are received, all of them are counted. An improved estimate of lost packets can be calculated by adding packetsDuplicated to packetsLost; this will always result in a positive number, but not the same number as RFC 3550 would calculate.
| UINT64 RtcInboundRtpStreamStats::packetsFailedDecryption |
The cumulative number of RTP packets that failed to be decrypted according to the procedures in [RFC3711]. These packets are not counted by packetsDiscarded.
| UINT32 RtcInboundRtpStreamStats::pliCount |
TODO Only valid for video. Count the total number of Picture Loss Indication (PLI) packets sent by this receiver.
| UINT64 RtcInboundRtpStreamStats::qpSum |
TODO Only valid for video. The sum of the QP values of frames decoded by this receiver. The count of frames is in framesDecoded. The definition of QP value depends on the codec; for VP8, the QP value is the value carried in the frame header as the syntax element "y_ac_qi", and defined in [RFC6386] section 19.2. Its range is 0..127. Note that the QP value is only an indication of quantizer values used; many formats have ways to vary the quantizer value within the frame.
| RtcReceivedRtpStreamStats RtcInboundRtpStreamStats::received |
| DOMString RtcInboundRtpStreamStats::receiverId |
TODO The stats ID used to look up the RTCAudioReceiverStats or RTCVideoReceiverStats object receiving this stream.
| DOMString RtcInboundRtpStreamStats::remoteId |
TODO The remoteId is used for looking up the remote RTCRemoteOutboundRtpStreamStats object for the same SSRC.
| UINT64 RtcInboundRtpStreamStats::removedSamplesForAcceleration |
TODO Only valid for audio. When playout is sped up, this counter is increased by the difference between the number of samples received and the number of samples played out.
| UINT64 RtcInboundRtpStreamStats::samplesDecodedWithCelt |
TODO Only valid for audio and when the audio codec is Opus. The total number of samples decoded by the CELT portion of the Opus codec.
| UINT64 RtcInboundRtpStreamStats::samplesDecodedWithSilk |
TODO Only valid for audio and when the audio codec is Opus. The total number of samples decoded by the SILK portion of the Opus codec.
| UINT64 RtcInboundRtpStreamStats::silentConcealedSamples |
TODO Only valid for audio. The total number of concealed samples inserted that are "silent".
| UINT32 RtcInboundRtpStreamStats::sliCount |
TODO Only valid for video. Count the total number of Slice Loss Indication (SLI) packets sent by this receiver.
| DOUBLE RtcInboundRtpStreamStats::totalAudioEnergy |
TODO Only valid for audio. Represents the audio energy of the receiving track. For audio energy of tracks attached locally, see RTCAudioSourceStats instead.
| DOUBLE RtcInboundRtpStreamStats::totalDecodeTime |
TODO Total number of seconds that have been spent decoding the framesDecoded frames of this stream. The average decode time can be calculated by dividing this value with framesDecoded. The time it takes to decode one frame is the time passed between feeding the decoder a frame and the decoder returning decoded data for that frame.
| DOUBLE RtcInboundRtpStreamStats::totalInterFrameDelay |
TODO Sum of the interframe delays in seconds between consecutively decoded frames, recorded just after a frame has been decoded. The interframe delay variance be calculated from totalInterFrameDelay, totalSquaredInterFrameDelay, and framesDecoded according to the formula: (totalSquaredInterFrameDelay - totalInterFrameDelay^2/ framesDecoded)/framesDecoded.
| DOUBLE RtcInboundRtpStreamStats::totalSamplesDuration |
TODO Only valid for audio. Represents the audio duration of the receiving track. For audio durations of tracks attached locally, see RTCAudioSourceStats instead.
| UINT64 RtcInboundRtpStreamStats::totalSamplesReceived |
TODO Only valid for audio. The total number of samples that have been received on this RTP stream. This includes concealedSamples.
| DOUBLE RtcInboundRtpStreamStats::totalSquaredInterFrameDelay |
TODO Sum of the squared interframe delays in seconds between consecutively decoded frames, recorded just after a frame has been decoded. See totalInterFrameDelay for details on how to calculate the interframe delay variance.
| DOMString RtcInboundRtpStreamStats::trackId |
TODO The identifier of the stats object representing the receiving track, an RTCReceiverAudioTrackAttachmentStats or RTCReceiverVideoTrackAttachmentStats.
| BOOL RtcInboundRtpStreamStats::voiceActivityFlag |
TODO Only valid for audio. Whether the last RTP packet whose frame was delivered to the RTCRtpReceiver's MediaStreamTrack for playout contained voice activity or not based on the presence of the V bit in the extension header, as defined in [RFC6464]. This is the stats-equivalent of RTCRtpSynchronizationSource.voiceActivityFlag in [[WEBRTC].
1.9.1