commit | 9e117c5e1b279aa2920786351eaf72b4f07ce295 | [log] [tgz] |
---|---|---|
author | stefan <stefan@webrtc.org> | Wed Aug 16 15:16:25 2017 |
committer | Commit Bot <commit-bot@chromium.org> | Wed Aug 16 15:16:25 2017 |
tree | 23c6983c077cecdf8fb96731c9a225390a452223 | |
parent | 5901d9dfe072e9e4b13459e122b26f6ff327fe78 [diff] |
Reland of Add functionality which limits the number of bytes on the network. (patchset #1 id:1 of https://codereview.webrtc.org/3001653002/ ) Reason for revert: Reland Original issue's description: > Revert of Add functionality which limits the number of bytes on the network. (patchset #26 id:500001 of https://codereview.webrtc.org/2918323002/ ) > > Reason for revert: > Speculative revert to see if this caused regressions in android perf tests. > > Original issue's description: > > Add functionality which limits the number of bytes on the network. > > > > The limit is based on the bandwidth delay product, but also adds some additional slack to compensate for the sawtooth-like BWE pattern and the slowness of the encoder rate control. The delay is estimated based on the time from sending a packet until an ack is received. Since acks are received in bursts (feedback is only sent periodically), a min filter is used to estimate the rtt. > > > > Whenever the in flight bytes reaches the congestion window, the pacer is paused, which in turn will result in send-side queues growing. Eventually the encoders will be paused as the pacer queue grows large (currently 2 seconds). > > > > BUG=webrtc:7926 > > > > Review-Url: https://codereview.webrtc.org/2918323002 > > Cr-Commit-Position: refs/heads/master@{#19289} > > Committed: https://chromium.googlesource.com/external/webrtc/+/8497fdde43d920ab1f0cc90362534e5493d23abe > > TBR=terelius@webrtc.org,philipel@webrtc.org,tschumim@webrtc.org,gnish@webrtc.org > # Not skipping CQ checks because original CL landed more than 1 days ago. > BUG=webrtc:7926 > > Review-Url: https://codereview.webrtc.org/3001653002 > Cr-Commit-Position: refs/heads/master@{#19339} > Committed: https://chromium.googlesource.com/external/webrtc/+/64136af364d1fecada49e35b1bfa39ef2641d5d0 TBR=terelius@webrtc.org,philipel@webrtc.org,tschumim@webrtc.org,gnish@webrtc.org # Not skipping CQ checks because original CL landed more than 1 days ago. BUG=webrtc:7926 Review-Url: https://codereview.webrtc.org/2994343002 Cr-Commit-Position: refs/heads/master@{#19373}
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.