0359ba2225
implements a total frame assembly time statistic that measures the cumulative time between the arrival of the first packet of a frame (the lowest reception time) and the time all packets of the frame have been received (i.e. the highest reception time) This is similar to totalProcessingDelay https://w3c.github.io/webrtc-stats/#dom-rtcinboundrtpstreamstats-totalprocessingdelay in particular with respect to only being incremented for frames that are being decoded but does not include the amount of time spent decoding the frame. This statistic is useful for evaluating mechanisms like NACK and FEC and gives some insight into the behavior of the pacer sending the packets. Note that for frames with just a single packet the assembly time will be zero. In order to calculate an average assembly time an additional frames_assembled_from_multiple_packets counter for frames with more than a single packet is added. Currently this is a nonstandard stat so will only show up in webrtc-internals and not in getStats. Formally it can be defined as totalAssemblyTime of type double Only exists for video. The sum of the time, in seconds, each video frame takes from the time the first RTP packet is received (reception timestamp) and to the time the last RTP packet of a frame is received. Given the complexities involved, the time of arrival or the reception timestamp is measured as close to the network layer as possible. This metric is not incremented for frames that are not decoded, i.e., framesDropped, partialFramesLost or frames that fail decoding for other reasons (if any). Only incremented for frames consisting of more than one RTP packet. The average frame assembly time can be calculated by dividing the totalAssemblyTime with framesAssembledFromMultiplePacket. framesAssembledFromMultiplePacket of type unsigned long Only exists for video. It represents the total number of frames correctly decoded for this RTP stream that consist of more than one RTP packet. For such frames the totalAssemblyTime is incremented. BUG=webrtc:13986 Change-Id: Ie0ae431d72a57a0001c3240daba8eda35955f04e Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/260920 Reviewed-by: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Henrik Boström <hbos@webrtc.org> Commit-Queue: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36922}
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.
Development
See here for instructions on how to get started developing with the native code.
Authoritative list of directories that contain the native API header files.
More info
- Official web site: http://www.webrtc.org
- Master source code repo: https://webrtc.googlesource.com/src
- Samples and reference apps: https://github.com/webrtc
- Mailing list: http://groups.google.com/group/discuss-webrtc
- Continuous build: https://ci.chromium.org/p/webrtc/g/ci/console
- Coding style guide
- Code of conduct
- Reporting bugs
- Documentation