commit | d81992197c9e31d9009a0c6ee9d3862479773c06 | [log] [tgz] |
---|---|---|
author | Henrik Boström <hbos@webrtc.org> | Wed Oct 26 14:50:53 2022 |
committer | WebRTC LUCI CQ <webrtc-scoped@luci-project-accounts.iam.gserviceaccount.com> | Wed Oct 26 21:29:20 2022 |
tree | 1e94d308d39a64472f37f820f0230e272b796dcf | |
parent | c34a8c19c6319bea140a8843e2494d69010d1f4f [diff] |
[Stats] Update totalPacketSendDelay to only cover time in pacer queue. This metric was always supposed to be the spec's answer to googBucketDelay, and is defined as "The total number of seconds that packets have spent buffered locally before being transmitted onto the network." But our implementation measured the time between capture and send, including encode time. This is incorrect and yields a much larger value than expected. This CL updated the metric to do what the spec says. Implementation-wise we measure the time between pushing and popping each packet from the queue (in modules/pacing/prioritized_packet_queue.cc). The spec says to increment the delay counter at the same time as we increment the packet counter in order for the app to be able to do "delta totalPacketSendDelay / delta packetSent". For this reason, `total_packet_delay` is added to RtpPacketCounter. (Previously, the two counters were incremented on different threads and observers.) Running Google Meet on a good network, I could observe a 2-3 ms average send delay per packet with this implementation compared to 20-30 ms with the old implementation. See b/137014977#comment170 for comparison with googBucketDelay which is a little bit different by design - totalPacketSendDelay is clearly better than googBucketDelay. Since none of this depend on the media kind, we can wire up this metric for audio as well in a follow-up: https://webrtc-review.googlesource.com/c/src/+/280523 Bug: webrtc:14593 Change-Id: If8fcd82fee74030d0923ee5df2c2aea2264600d4 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/280443 Reviewed-by: Erik Språng <sprang@webrtc.org> Commit-Queue: Henrik Boström <hbos@webrtc.org> Cr-Commit-Position: refs/heads/main@{#38480}
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 here for instructions on how to get started developing with the native code.
Authoritative list of directories that contain the native API header files.