b0ed12099f
As part of go/coil update code search links to not point to the "master" branch. Bug: chromium:1226942 Change-Id: I0ae9e84ecc660f789a69fe0b226f93bbc39a8a66 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/226081 Commit-Queue: Tony Herre <toprice@chromium.org> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/master@{#34531}
218 lines
16 KiB
Markdown
218 lines
16 KiB
Markdown
<?% config.freshness.reviewed = '2021-04-14' %?>
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<?% config.freshness.owner = 'asapersson' %?>
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# Video stats
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Overview of collected statistics for [VideoSendStream] and [VideoReceiveStream].
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## VideoSendStream
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[VideoSendStream::Stats] for a sending stream can be gathered via `VideoSendStream::GetStats()`.
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Some statistics are collected per RTP stream (see [StreamStats]) and can be of `StreamType`: `kMedia`, `kRtx`, `kFlexfec`.
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Multiple `StreamStats` objects are for example present if simulcast is used (multiple `kMedia` objects) or if RTX or FlexFEC is negotiated.
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### SendStatisticsProxy
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`VideoSendStream` owns a [SendStatisticsProxy] which implements
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`VideoStreamEncoderObserver`,
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`RtcpStatisticsCallback`,
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`ReportBlockDataObserver`,
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`RtcpPacketTypeCounterObserver`,
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`StreamDataCountersCallback`,
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`BitrateStatisticsObserver`,
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`FrameCountObserver`,
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`SendSideDelayObserver`
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and holds a `VideoSendStream::Stats` object.
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`SendStatisticsProxy` is called via these interfaces by different components (e.g. `RtpRtcp` module) to update stats.
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#### StreamStats
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* `type` - kMedia, kRtx or kFlexfec.
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* `referenced_media_ssrc` - only present for type kRtx/kFlexfec. The SSRC for the kMedia stream that retransmissions or FEC is performed for.
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Updated when a frame has been encoded, `VideoStreamEncoder::OnEncodedImage`.
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* `frames_encoded `- total number of encoded frames.
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* `encode_frame_rate` - number of encoded frames during the last second.
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* `width` - width of last encoded frame [[rtcoutboundrtpstreamstats-framewidth]].
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* `height` - height of last encoded frame [[rtcoutboundrtpstreamstats-frameheight]].
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* `total_encode_time_ms` - total encode time for encoded frames.
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* `qp_sum` - sum of quantizer values of encoded frames [[rtcoutboundrtpstreamstats-qpsum]].
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* `frame_counts` - total number of encoded key/delta frames [[rtcoutboundrtpstreamstats-keyframesencoded]].
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Updated when a RTP packet is transmitted to the network, `RtpSenderEgress::SendPacket`.
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* `rtp_stats` - total number of sent bytes/packets.
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* `total_bitrate_bps` - total bitrate sent in bits per second (over a one second window).
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* `retransmit_bitrate_bps` - total retransmit bitrate sent in bits per second (over a one second window).
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* `avg_delay_ms` - average capture-to-send delay for sent packets (over a one second window).
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* `max_delay_ms` - maximum capture-to-send delay for sent packets (over a one second window).
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* `total_packet_send_delay_ms` - total capture-to-send delay for sent packets [[rtcoutboundrtpstreamstats-totalpacketsenddelay]].
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Updated when an incoming RTCP packet is parsed, `RTCPReceiver::ParseCompoundPacket`.
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* `rtcp_packet_type_counts` - total number of received NACK/FIR/PLI packets [rtcoutboundrtpstreamstats-[nackcount], [fircount], [plicount]].
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Updated when a RTCP report block packet is received, `RTCPReceiver::TriggerCallbacksFromRtcpPacket`.
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* `rtcp_stats` - RTCP report block data.
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* `report_block_data` - RTCP report block data.
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#### Stats
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* `std::map<uint32_t, StreamStats> substreams` - StreamStats mapped per SSRC.
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Updated when a frame is received from the source, `VideoStreamEncoder::OnFrame`.
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* `frames` - total number of frames fed to VideoStreamEncoder.
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* `input_frame_rate` - number of frames fed to VideoStreamEncoder during the last second.
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* `frames_dropped_by_congestion_window` - total number of dropped frames due to congestion window pushback.
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* `frames_dropped_by_encoder_queue` - total number of dropped frames due to that the encoder is blocked.
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Updated if a frame from the source is dropped, `VideoStreamEncoder::OnDiscardedFrame`.
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* `frames_dropped_by_capturer` - total number dropped frames by the source.
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Updated if a frame is dropped by `FrameDropper`, `VideoStreamEncoder::MaybeEncodeVideoFrame`.
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* `frames_dropped_by_rate_limiter` - total number of dropped frames to avoid bitrate overuse.
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Updated (if changed) before a frame is passed to the encoder, `VideoStreamEncoder::EncodeVideoFrame`.
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* `encoder_implementation_name` - name of encoder implementation [[rtcoutboundrtpstreamstats-encoderimplementation]].
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Updated after a frame has been encoded, `VideoStreamEncoder::OnEncodedImage`.
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* `frames_encoded `- total number of encoded frames [[rtcoutboundrtpstreamstats-framesencoded]].
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* `encode_frame_rate` - number of encoded frames during the last second [[rtcoutboundrtpstreamstats-framespersecond]].
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* `total_encoded_bytes_target` - total target frame size in bytes [[rtcoutboundrtpstreamstats-totalencodedbytestarget]].
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* `huge_frames_sent` - total number of huge frames sent [[rtcoutboundrtpstreamstats-hugeframessent]].
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* `media_bitrate_bps` - the actual bitrate the encoder is producing.
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* `avg_encode_time_ms` - average encode time for encoded frames.
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* `total_encode_time_ms` - total encode time for encoded frames [[rtcoutboundrtpstreamstats-totalencodetime]].
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* `frames_dropped_by_encoder`- total number of dropped frames by the encoder.
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Adaptation stats.
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* `bw_limited_resolution` - shows if resolution is limited due to restricted bandwidth.
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* `cpu_limited_resolution` - shows if resolution is limited due to cpu.
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* `bw_limited_framerate` - shows if framerate is limited due to restricted bandwidth.
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* `cpu_limited_framerate` - shows if framerate is limited due to cpu.
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* `quality_limitation_reason` - current reason for limiting resolution and/or framerate [[rtcoutboundrtpstreamstats-qualitylimitationreason]].
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* `quality_limitation_durations_ms` - total time spent in quality limitation state [[rtcoutboundrtpstreamstats-qualitylimitationdurations]].
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* `quality_limitation_resolution_changes` - total number of times that resolution has changed due to quality limitation [[rtcoutboundrtpstreamstats-qualitylimitationresolutionchanges]].
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* `number_of_cpu_adapt_changes` - total number of times resolution/framerate has changed due to cpu limitation.
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* `number_of_quality_adapt_changes` - total number of times resolution/framerate has changed due to quality limitation.
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Updated when the encoder is configured, `VideoStreamEncoder::ReconfigureEncoder`.
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* `content_type` - configured content type (UNSPECIFIED/SCREENSHARE).
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Updated when the available bitrate changes, `VideoSendStreamImpl::OnBitrateUpdated`.
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* `target_media_bitrate_bps` - the bitrate the encoder is configured to use.
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* `suspended` - shows if video is suspended due to zero target bitrate.
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## VideoReceiveStream
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[VideoReceiveStream::Stats] for a receiving stream can be gathered via `VideoReceiveStream::GetStats()`.
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### ReceiveStatisticsProxy
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`VideoReceiveStream` owns a [ReceiveStatisticsProxy] which implements
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`VCMReceiveStatisticsCallback`,
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`RtcpCnameCallback`,
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`RtcpPacketTypeCounterObserver`,
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`CallStatsObserver`
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and holds a `VideoReceiveStream::Stats` object.
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`ReceiveStatisticsProxy` is called via these interfaces by different components (e.g. `RtpRtcp` module) to update stats.
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#### Stats
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* `current_payload_type` - current payload type.
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* `ssrc` - configured SSRC for the received stream.
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Updated when a complete frame is received, `FrameBuffer::InsertFrame`.
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* `frame_counts` - total number of key/delta frames received [[rtcinboundrtpstreamstats-keyframesdecoded]].
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* `network_frame_rate` - number of frames received during the last second.
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Updated when a frame is ready for decoding, `FrameBuffer::GetNextFrame`. From `VCMTiming`:
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* `jitter_buffer_ms` - jitter buffer delay in ms.
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* `max_decode_ms` - the 95th percentile observed decode time within a time window (10 sec).
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* `render_delay_ms` - render delay in ms.
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* `min_playout_delay_ms` - minimum playout delay in ms.
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* `target_delay_ms` - target playout delay in ms. Max(`min_playout_delay_ms`, `jitter_delay_ms` + `max_decode_ms` + `render_delay_ms`).
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* `current_delay_ms` - actual playout delay in ms.
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* `jitter_buffer_delay_seconds` - total jitter buffer delay in seconds [[rtcinboundrtpstreamstats-jitterbufferdelay]].
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* `jitter_buffer_emitted_count` - total number of frames that have come out from the jitter buffer [[rtcinboundrtpstreamstats-jitterbufferemittedcount]].
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Updated (if changed) after a frame is passed to the decoder, `VCMGenericDecoder::Decode`.
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* `decoder_implementation_name` - name of decoder implementation [[rtcinboundrtpstreamstats-decoderimplementation]].
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Updated when a frame is ready for decoding, `FrameBuffer::GetNextFrame`.
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* `timing_frame_info` - timestamps for a full lifetime of a frame.
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* `first_frame_received_to_decoded_ms` - initial decoding latency between the first arrived frame and the first decoded frame.
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* `frames_dropped` - total number of dropped frames prior to decoding or if the system is too slow [[rtcreceivedrtpstreamstats-framesdropped]].
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Updated after a frame has been decoded, `VCMDecodedFrameCallback::Decoded`.
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* `frames_decoded` - total number of decoded frames [[rtcinboundrtpstreamstats-framesdecoded]].
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* `decode_frame_rate` - number of decoded frames during the last second [[rtcinboundrtpstreamstats-framespersecond]].
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* `decode_ms` - time to decode last frame in ms.
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* `total_decode_time_ms` - total decode time for decoded frames [[rtcinboundrtpstreamstats-totaldecodetime]].
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* `qp_sum` - sum of quantizer values of decoded frames [[rtcinboundrtpstreamstats-qpsum]].
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* `content_type` - content type (UNSPECIFIED/SCREENSHARE).
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* `interframe_delay_max_ms` - max inter-frame delay within a time window between decoded frames.
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* `total_inter_frame_delay` - sum of inter-frame delay in seconds between decoded frames [[rtcinboundrtpstreamstats-totalinterframedelay]].
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* `total_squared_inter_frame_delay` - sum of squared inter-frame delays in seconds between decoded frames [[rtcinboundrtpstreamstats-totalsquaredinterframedelay]].
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Updated before a frame is sent to the renderer, `VideoReceiveStream2::OnFrame`.
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* `frames_rendered` - total number of rendered frames.
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* `render_frame_rate` - number of rendered frames during the last second.
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* `width` - width of last frame fed to renderer [[rtcinboundrtpstreamstats-framewidth]].
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* `height` - height of last frame fed to renderer [[rtcinboundrtpstreamstats-frameheight]].
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* `estimated_playout_ntp_timestamp_ms` - estimated playout NTP timestamp [[rtcinboundrtpstreamstats-estimatedplayouttimestamp]].
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* `sync_offset_ms` - NTP timestamp difference between the last played out audio and video frame.
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* `freeze_count` - total number of detected freezes.
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* `pause_count` - total number of detected pauses.
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* `total_freezes_duration_ms` - total duration of freezes in ms.
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* `total_pauses_duration_ms` - total duration of pauses in ms.
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* `total_frames_duration_ms` - time in ms between the last rendered frame and the first rendered frame.
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* `sum_squared_frame_durations` - sum of squared inter-frame delays in seconds between rendered frames.
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`ReceiveStatisticsImpl::OnRtpPacket` is updated for received RTP packets. From `ReceiveStatistics`:
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* `total_bitrate_bps` - incoming bitrate in bps.
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* `rtp_stats` - RTP statistics for the received stream.
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Updated when a RTCP packet is sent, `RTCPSender::ComputeCompoundRTCPPacket`.
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* `rtcp_packet_type_counts` - total number of sent NACK/FIR/PLI packets [rtcinboundrtpstreamstats-[nackcount], [fircount], [plicount]].
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[VideoSendStream]: https://source.chromium.org/chromium/chromium/src/+/main:third_party/webrtc/call/video_send_stream.h
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[VideoSendStream::Stats]: https://source.chromium.org/chromium/chromium/src/+/main:third_party/webrtc/call/video_send_stream.h?q=VideoSendStream::Stats
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[StreamStats]: https://source.chromium.org/chromium/chromium/src/+/main:third_party/webrtc/call/video_send_stream.h?q=VideoSendStream::StreamStats
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[SendStatisticsProxy]: https://source.chromium.org/chromium/chromium/src/+/main:third_party/webrtc/video/send_statistics_proxy.h
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[rtcoutboundrtpstreamstats-framewidth]: https://w3c.github.io/webrtc-stats/#dom-rtcoutboundrtpstreamstats-framewidth
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[rtcoutboundrtpstreamstats-frameheight]: https://w3c.github.io/webrtc-stats/#dom-rtcoutboundrtpstreamstats-frameheight
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[rtcoutboundrtpstreamstats-qpsum]: https://w3c.github.io/webrtc-stats/#dom-rtcoutboundrtpstreamstats-qpsum
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[rtcoutboundrtpstreamstats-keyframesencoded]: https://w3c.github.io/webrtc-stats/#dom-rtcoutboundrtpstreamstats-keyframesencoded
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[rtcoutboundrtpstreamstats-totalpacketsenddelay]: https://w3c.github.io/webrtc-stats/#dom-rtcoutboundrtpstreamstats-totalpacketsenddelay
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[nackcount]: https://w3c.github.io/webrtc-stats/#dom-rtcoutboundrtpstreamstats-nackcount
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[fircount]: https://w3c.github.io/webrtc-stats/#dom-rtcoutboundrtpstreamstats-fircount
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[plicount]: https://w3c.github.io/webrtc-stats/#dom-rtcoutboundrtpstreamstats-plicount
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[rtcoutboundrtpstreamstats-encoderimplementation]: https://w3c.github.io/webrtc-stats/#dom-rtcoutboundrtpstreamstats-encoderimplementation
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[rtcoutboundrtpstreamstats-framesencoded]: https://w3c.github.io/webrtc-stats/#dom-rtcoutboundrtpstreamstats-framesencoded
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[rtcoutboundrtpstreamstats-framespersecond]: https://w3c.github.io/webrtc-stats/#dom-rtcoutboundrtpstreamstats-framespersecond
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[rtcoutboundrtpstreamstats-totalencodedbytestarget]: https://w3c.github.io/webrtc-stats/#dom-rtcoutboundrtpstreamstats-totalencodedbytestarget
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[rtcoutboundrtpstreamstats-hugeframessent]: https://w3c.github.io/webrtc-stats/#dom-rtcoutboundrtpstreamstats-hugeframessent
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[rtcoutboundrtpstreamstats-totalencodetime]: https://w3c.github.io/webrtc-stats/#dom-rtcoutboundrtpstreamstats-totalencodetime
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[rtcoutboundrtpstreamstats-qualitylimitationreason]: https://w3c.github.io/webrtc-stats/#dom-rtcoutboundrtpstreamstats-qualitylimitationreason
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[rtcoutboundrtpstreamstats-qualitylimitationdurations]: https://w3c.github.io/webrtc-stats/#dom-rtcoutboundrtpstreamstats-qualitylimitationdurations
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[rtcoutboundrtpstreamstats-qualitylimitationresolutionchanges]: https://w3c.github.io/webrtc-stats/#dom-rtcoutboundrtpstreamstats-qualitylimitationresolutionchanges
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[VideoReceiveStream]: https://source.chromium.org/chromium/chromium/src/+/main:third_party/webrtc/call/video_receive_stream.h
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[VideoReceiveStream::Stats]: https://source.chromium.org/chromium/chromium/src/+/main:third_party/webrtc/call/video_receive_stream.h?q=VideoReceiveStream::Stats
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[ReceiveStatisticsProxy]: https://source.chromium.org/chromium/chromium/src/+/main:third_party/webrtc/video/receive_statistics_proxy2.h
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[rtcinboundrtpstreamstats-keyframesdecoded]: https://w3c.github.io/webrtc-stats/#dom-rtcinboundrtpstreamstats-keyframesdecoded
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[rtcinboundrtpstreamstats-jitterbufferdelay]: https://w3c.github.io/webrtc-stats/#dom-rtcinboundrtpstreamstats-jitterbufferdelay
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[rtcinboundrtpstreamstats-jitterbufferemittedcount]: https://w3c.github.io/webrtc-stats/#dom-rtcinboundrtpstreamstats-jitterbufferemittedcount
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[rtcinboundrtpstreamstats-decoderimplementation]: https://w3c.github.io/webrtc-stats/#dom-rtcinboundrtpstreamstats-decoderimplementation
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[rtcreceivedrtpstreamstats-framesdropped]: https://www.w3.org/TR/webrtc-stats/#dom-rtcreceivedrtpstreamstats-framesdropped
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[rtcinboundrtpstreamstats-framesdecoded]: https://w3c.github.io/webrtc-stats/#dom-rtcinboundrtpstreamstats-framesdecoded
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[rtcinboundrtpstreamstats-framespersecond]: https://w3c.github.io/webrtc-stats/#dom-rtcinboundrtpstreamstats-framespersecond
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[rtcinboundrtpstreamstats-totaldecodetime]: https://w3c.github.io/webrtc-stats/#dom-rtcinboundrtpstreamstats-totaldecodetime
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[rtcinboundrtpstreamstats-qpsum]: https://w3c.github.io/webrtc-stats/#dom-rtcinboundrtpstreamstats-qpsum
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[rtcinboundrtpstreamstats-totalinterframedelay]: https://w3c.github.io/webrtc-stats/#dom-rtcinboundrtpstreamstats-totalinterframedelay
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[rtcinboundrtpstreamstats-totalsquaredinterframedelay]: https://w3c.github.io/webrtc-stats/#dom-rtcinboundrtpstreamstats-totalsquaredinterframedelay
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[rtcinboundrtpstreamstats-estimatedplayouttimestamp]: https://w3c.github.io/webrtc-stats/#dom-rtcinboundrtpstreamstats-estimatedplayouttimestamp
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[rtcinboundrtpstreamstats-framewidth]: https://w3c.github.io/webrtc-stats/#dom-rtcinboundrtpstreamstats-framewidth
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[rtcinboundrtpstreamstats-frameheight]: https://w3c.github.io/webrtc-stats/#dom-rtcinboundrtpstreamstats-frameheight
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[nackcount]: https://w3c.github.io/webrtc-stats/#dom-rtcinboundrtpstreamstats-nackcount
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[fircount]: https://w3c.github.io/webrtc-stats/#dom-rtcinboundrtpstreamstats-fircount
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[plicount]: https://w3c.github.io/webrtc-stats/#dom-rtcinboundrtpstreamstats-plicount
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