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webrtcbin: Update documentation of 'get-stats' action signal
Some stats fields are updated according to the current implementation. Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/2005>
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1 changed files with 15 additions and 7 deletions
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@ -7783,25 +7783,29 @@ gst_webrtc_bin_class_init (GstWebRTCBinClass * klass)
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* "ssrc" G_TYPE_STRING the rtp sequence src in use
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* "transport-id" G_TYPE_STRING identifier for the associated RTCTransportStats for this stream
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* "codec-id" G_TYPE_STRING identifier for the associated RTCCodecStats for this stream
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* "fir-count" G_TYPE_UINT FIR requests received by the sender (only for local statistics)
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* "pli-count" G_TYPE_UINT PLI requests received by the sender (only for local statistics)
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* "nack-count" G_TYPE_UINT NACK requests received by the sender (only for local statistics)
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*
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* RTCReceivedStreamStats supported fields (https://w3c.github.io/webrtc-stats/#receivedrtpstats-dict*)
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*
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* "packets-received" G_TYPE_UINT64 number of packets received (only for local inbound)
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* "bytes-received" G_TYPE_UINT64 number of bytes received (only for local inbound)
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* "packets-lost" G_TYPE_UINT number of packets lost
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* "jitter" G_TYPE_DOUBLE packet jitter measured in secondss
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* "packets-received" G_TYPE_UINT64 number of packets received (only for local inbound)
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* "packets-lost" G_TYPE_UINT64 number of packets lost
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* "packets-discarded" G_TYPE_UINT64 number of packets discarded
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* "packets-repaired" G_TYPE_UINT64 number of packets repaired
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* "jitter" G_TYPE_DOUBLE packet jitter measured in seconds
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*
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* RTCInboundRTPStreamStats supported fields (https://w3c.github.io/webrtc-stats/#inboundrtpstats-dict*)
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*
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* "remote-id" G_TYPE_STRING identifier for the associated RTCRemoteOutboundRTPStreamStats
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* "bytes-received" G_TYPE_UINT64 number of bytes received (only for local inbound)
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* "packets-duplicated" G_TYPE_UINT64 number of packets duplicated
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* "fir-count" G_TYPE_UINT FIR packets sent by the receiver
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* "pli-count" G_TYPE_UINT PLI packets sent by the receiver
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* "nack-count" G_TYPE_UINT NACK packets sent by the receiver
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*
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* RTCRemoteInboundRTPStreamStats supported fields (https://w3c.github.io/webrtc-stats/#remoteinboundrtpstats-dict*)
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*
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* "local-id" G_TYPE_STRING identifier for the associated RTCOutboundRTPSTreamStats
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* "round-trip-time" G_TYPE_DOUBLE round trip time of packets measured in seconds
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* "fraction-lost" G_TYPE_DOUBLE fraction packet loss
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*
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* RTCSentRTPStreamStats supported fields (https://w3c.github.io/webrtc-stats/#sentrtpstats-dict*)
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*
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@ -7811,10 +7815,14 @@ gst_webrtc_bin_class_init (GstWebRTCBinClass * klass)
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* RTCOutboundRTPStreamStats supported fields (https://w3c.github.io/webrtc-stats/#outboundrtpstats-dict*)
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*
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* "remote-id" G_TYPE_STRING identifier for the associated RTCRemoteInboundRTPSTreamStats
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* "fir-count" G_TYPE_UINT FIR packets received by the sender
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* "pli-count" G_TYPE_UINT PLI packets received by the sender
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* "nack-count" G_TYPE_UINT NACK packets received by the sender
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*
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* RTCRemoteOutboundRTPStreamStats supported fields (https://w3c.github.io/webrtc-stats/#remoteoutboundrtpstats-dict*)
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*
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* "local-id" G_TYPE_STRING identifier for the associated RTCInboundRTPSTreamStats
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* "remote-timestamp" G_TYPE_DOUBLE remote timestamp the statistics were sent by the remote
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*
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*/
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gst_webrtc_bin_signals[GET_STATS_SIGNAL] =
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