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