Adding upstream version 124.0.1.upstream/124.0.1

Signed-off-by: Daniel Baumann <daniel.baumann@progress-linux.org>

2 files changed, 329 insertions, 0 deletions

diff --git a/third_party/libwebrtc/video/g3doc/adaptation.md b/third_party/libwebrtc/video/g3doc/adaptation.md
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+<!-- go/cmark -->
+<!--* freshness: {owner: 'eshr' reviewed: '2021-04-13'} *-->
+
+# Video Adaptation
+
+Video adaptation is a mechanism which reduces the bandwidth or CPU consumption
+by reducing encoded video quality.
+
+## Overview
+
+Adaptation occurs when a _Resource_ signals that it is currently underused or
+overused. When overused, the video quality is decreased and when underused, the
+video quality is increased. There are currently two dimensions in which the
+quality can be adapted: frame-rate and resolution. The dimension that is adapted
+is based on the degradation preference for the video track.
+
+## Resources
+
+_Resources_ monitor metrics from the system or the video stream. For example, a
+resource could monitor system temperature or the bandwidth usage of the video
+stream. A resource implements the [Resource][resource.h] interface. When a
+resource detects that it is overused, it calls `SetUsageState(kOveruse)`. When
+the resource is no longer overused, it can signal this using
+`SetUsageState(kUnderuse)`.
+
+There are two resources that are used by default on all video tracks: Quality
+scaler resource and encode overuse resource.
+
+### QP Scaler Resource
+
+The quality scaler resource monitors the quantization parameter (QP) of the
+encoded video frames for video send stream and ensures that the quality of the
+stream is acceptable for the current resolution. After each frame is encoded the
+[QualityScaler][quality_scaler.h] is given the QP of the encoded frame. Overuse
+or underuse is signalled when the average QP is outside of the
+[QP thresholds][VideoEncoder::QpThresholds]. If the average QP is above the
+_high_ threshold, the QP scaler signals _overuse_, and when below the _low_
+threshold the QP scaler signals _underuse_.
+
+The thresholds are set by the video encoder in the `scaling_settings` property
+of the [EncoderInfo][EncoderInfo].
+
+*Note:* that the QP scaler is only enabled when the degradation preference is
+`MAINTAIN_FRAMERATE` or `BALANCED`.
+
+### Encode Usage Resource
+
+The [encoder usage resource][encode_usage_resource.h] monitors how long it takes
+to encode a video frame. This works as a good proxy measurement for CPU usage as
+contention increases when CPU usage is high, increasing the encode times of the
+video frames.
+
+The time is tracked from when frame encoding starts to when it is completed. If
+the average encoder usage exceeds the thresholds set, *overuse* is triggered.
+
+### Injecting other Resources
+
+A custom resource can be injected into the call using the
+[Call::AddAdaptationResource][Call::AddAdaptationResource] method.
+
+## Adaptation
+
+When a a *resource* signals the it is over or underused, this signal reaches the
+`ResourceAdaptationProcessor` who requests an `Adaptation` proposal from the
+[VideoStreamAdapter][VideoStreamAdapter]. This proposal is based on the
+degradation preference of the video stream. `ResourceAdaptationProcessor` will
+determine if the `Adaptation` should be applied based on the current adaptation
+status and the `Adaptation` proposal.
+
+### Degradation Preference
+
+There are 3 degradation preferences, described in the
+[RtpParameters][RtpParameters] header. These are
+
+*   `MAINTIAIN_FRAMERATE`: Adapt video resolution
+*   `MAINTIAIN_RESOLUTION`: Adapt video frame-rate.
+*   `BALANCED`: Adapt video frame-rate or resolution.
+
+The degradation preference is set for a video track using the
+`degradation_preference` property in the [RtpParameters][RtpParameters].
+
+## VideoSinkWants and video stream adaptation
+
+Once an adaptation is applied it notifies the video stream. The video stream
+converts this adaptation to a [VideoSinkWants][VideoSinkWants]. These sink wants
+indicate to the video stream that some restrictions should be applied to the
+stream before it is sent to encoding. It has a few properties, but for
+adaptation the properties that might be set are:
+
+*   `target_pixel_count`: The desired number of pixels for each video frame. The
+    actual pixel count should be close to this but does not have to be exact so
+    that aspect ratio can be maintained.
+*   `max_pixel_count`: The maximum number of pixels in each video frame. This
+    value can not be exceeded if set.
+*   `max_framerate_fps`: The maximum frame-rate for the video source. The source
+    is expected to drop frames that cause this threshold to be exceeded.
+
+The `VideoSinkWants` can be applied by any video source, or one may use the
+[AdaptedVideoTraceSource][adapted_video_track_source.h] which is a base class
+for sources that need video adaptation.
+
+[RtpParameters]: https://source.chromium.org/chromium/chromium/src/+/main:third_party/webrtc/api/rtp_parameters.h?q=%22RTC_EXPORT%20RtpParameters%22
+[resource.h]: https://source.chromium.org/chromium/chromium/src/+/main:third_party/webrtc/api/adaptation/resource.h
+[Call::AddAdaptationResource]: https://source.chromium.org/chromium/chromium/src/+/main:third_party/webrtc/call/call.h?q=Call::AddAdaptationResource
+[quality_scaler.h]: https://source.chromium.org/chromium/chromium/src/+/main:third_party/webrtc/modules/video_coding/utility/quality_scaler.h
+[VideoEncoder::QpThresholds]: https://source.chromium.org/chromium/chromium/src/+/main:third_party/webrtc/api/video_codecs/video_encoder.h?q=VideoEncoder::QpThresholds
+[EncoderInfo]: https://source.chromium.org/chromium/chromium/src/+/main:third_party/webrtc/api/video_codecs/video_encoder.h?q=VideoEncoder::EncoderInfo
+[encode_usage_resource.h]: https://source.chromium.org/chromium/chromium/src/+/main:third_party/webrtc/video/adaptation/encode_usage_resource.h
+[VideoStreamAdapter]: https://source.chromium.org/chromium/chromium/src/+/main:third_party/webrtc/call/adaptation/video_stream_adapter.h
+[adaptation_constraint.h]: https://source.chromium.org/chromium/chromium/src/+/main:third_party/webrtc/call/adaptation/adaptation_constraint.h
+[bitrate_constraint.h]: https://source.chromium.org/chromium/chromium/src/+/main:third_party/webrtc/video/adaptation/bitrate_constraint.h
+[AddOrUpdateSink]: https://source.chromium.org/chromium/chromium/src/+/main:third_party/webrtc/api/video/video_source_interface.h?q=AddOrUpdateSink
+[VideoSinkWants]: https://source.chromium.org/chromium/chromium/src/+/main:third_party/webrtc/api/video/video_source_interface.h?q=%22RTC_EXPORT%20VideoSinkWants%22
+[adapted_video_track_source.h]: https://source.chromium.org/chromium/chromium/src/+/main:third_party/webrtc/media/base/adapted_video_track_source.h
diff --git a/third_party/libwebrtc/video/g3doc/stats.md b/third_party/libwebrtc/video/g3doc/stats.md
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+<!-- go/cmark -->
+<!--* freshness: {owner: 'asapersson' reviewed: '2021-04-14'} *-->
+
+# 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