commit | f45ca3787fa8ed326f2ecc28296cd7b86f238fe9 | [log] [tgz] |
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author | Henrik Boström <hbos@webrtc.org> | Tue Mar 24 12:30:50 2020 |
committer | Commit Bot <commit-bot@chromium.org> | Tue Mar 24 13:31:54 2020 |
tree | 41f7adfd5c201060997ffaf1078d6fcfac60c330 | |
parent | 536c39539184ca0edb49393231193204da3868b4 [diff] |
[Stats] Explicit RTP-RTX and RTP-FEC mappings. Unblocks simulcast stats. --- Background --- The webrtc::VideoSendStream::StreamStats are converted into VideoSenderInfo objects which turn into "outbound-rtp" stats objects in getStats() (or "ssrc" objects in legacy getStats()). StreamStats are created for each type of substream: RTP media streams, RTX streams and FlexFEC streams - each with individual packet counters. The RTX stream is responsible for retransmissions of a referenced media stream and the FlexFEC stream is responsible for FEC of a referenced media stream. RTX/FEC streams do not show up as separate objects in getStats(). Only the media streams become "outbound-rtp" objects, but their packet and byte counters have to include the RTX and FEC counters. --- Overview of this CL --- This CL adds MergeInfoAboutOutboundRtpSubstreams(). It takes StreamStats of all kinds as input, and outputs media-only StreamStats - incorporating the RTX and FEC counters into the relevant media StreamStats. The merged StreamStats objects is a smaller set of objects than the non-merged counterparts, but when aggregating all packet counters together we end up with exact same packet and count as before. Because WebRtcVideoSendStream::GetVideoSenderInfo() currently aggregates the StreamStats into a single VideoSenderInfo (single "outbound-rtp"), this CL should not have any observable side-effects. Prior to this CL: aggregate StreamStats. After this CL: merge StreamStats and then aggregate them. However, when simulcast stats are implemented (WIP CL: https://webrtc-review.googlesource.com/c/src/+/168120) each RTP media stream should turn into an individual "outbound-rtp" object. We will then no longer aggregate all StreamStats into a single "info". This CL unblocks simulcast stats by providing StreamStats objects that could be turned into individual VideoSenderInfos. --- The Changes --- 1. Methods added to RtpConfig to be able to easily tell the relationship between RTP, RTX and FEC ssrcs. 2. StreamStats gets a StreamType (kMedia, kRtx or kFlexfec) that replaces the booleans (is_rtx, is_flexfec). 3. "referenced_media_ssrc" is added to StreamStats, making it possible to tell which kRtx/kFlexFec stream stats need to be merged with which kMedia StreamStats. 4. MergeInfoAboutOutboundRtpSubstreams() added and used. Bug: webrtc:11439 Change-Id: Iaf9002041169a054ddfd32c7ea06bd1dc36c6bca Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/170826 Commit-Queue: Henrik Boström <hbos@webrtc.org> Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Stefan Holmer <stefan@webrtc.org> Cr-Commit-Position: refs/heads/master@{#30869}
WebRTC is a free, open software project that provides browsers and mobile applications with Real-Time Communications (RTC) capabilities via simple APIs. The WebRTC components have been optimized to best serve this purpose.
Our mission: To enable rich, high-quality RTC applications to be developed for the browser, mobile platforms, and IoT devices, and allow them all to communicate via a common set of protocols.
The WebRTC initiative is a project supported by Google, Mozilla and Opera, amongst others.
See http://www.webrtc.org/native-code/development for instructions on how to get started developing with the native code.
Authoritative list of directories that contain the native API header files.