commit | 6bb5ab9740d22eacfbf7b4b5bd100a1b667b2911 | [log] [tgz] |
---|---|---|
author | Niels Möller <nisse@webrtc.org> | Fri Jan 11 10:11:10 2019 |
committer | Commit Bot <commit-bot@chromium.org> | Fri Jan 11 10:46:33 2019 |
tree | aa6ed913ca1a5b9d5d3cbb95a65b72e4777301bc | |
parent | 9c843906caff1ab4cc35b3026a9ca957fe34abe1 [diff] |
Reland "Refactor and remove media_optimization::MediaOptimization." This reverts commit 6613f8e98ab3654ade7e8f5352d8d6711b157499. Reason for revert: This change seemed innocent after all, so undoing speculative revert. Original change's description: > Revert "Refactor and remove media_optimization::MediaOptimization." > > This reverts commit 07276e4f89a93b1479d7aeefa53b4fc32daf001b. > > Reason for revert: Speculative revert due to downstream crashes. > > Original change's description: > > Refactor and remove media_optimization::MediaOptimization. > > > > This CL removes MediaOptmization and folds some of its functionality > > into VideoStreamEncoder. > > > > The FPS tracking is now handled by a RateStatistics instance. Frame > > dropping is still handled by FrameDropper. Both of these now live > > directly in VideoStreamEncoder. > > There is no intended change in behavior from this CL, but due to a new > > way of measuring frame rate, some minor perf changes can be expected. > > > > A small change in behavior is that OnBitrateUpdated is now called > > directly rather than on the next frame. Since both encoding frame and > > setting rate allocations happen on the encoder worker thread, there's > > really no reason to cache bitrates and wait until the next frame. > > An edge case though is that if a new bitrate is set before the first > > frame, we must remember that bitrate and then apply it after the video > > bitrate allocator has been first created. > > > > In addition to existing unit tests, manual tests have been used to > > confirm that frame dropping works as expected with misbehaving encoders. > > > > Bug: webrtc:10164 > > Change-Id: I7ee9c8d3c4f2bcf23c8c420310b05a4d35d94744 > > Reviewed-on: https://webrtc-review.googlesource.com/c/115620 > > Commit-Queue: Erik Språng <sprang@webrtc.org> > > Reviewed-by: Niels Moller <nisse@webrtc.org> > > Cr-Commit-Position: refs/heads/master@{#26147} > > TBR=nisse@webrtc.org,sprang@webrtc.org > > # Not skipping CQ checks because original CL landed > 1 day ago. > > Bug: webrtc:10164 > Change-Id: Ie0dae19dd012bc09e793c9661a45823fd760c20c > Reviewed-on: https://webrtc-review.googlesource.com/c/116780 > Reviewed-by: Niels Moller <nisse@webrtc.org> > Reviewed-by: Erik Språng <sprang@webrtc.org> > Commit-Queue: Niels Moller <nisse@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#26191} TBR=nisse@webrtc.org,sprang@webrtc.org Change-Id: Ieda1fad301de002460bb0bf5a75267ea065176a8 No-Presubmit: true No-Tree-Checks: true No-Try: true Bug: webrtc:10164 Reviewed-on: https://webrtc-review.googlesource.com/c/116960 Reviewed-by: Niels Moller <nisse@webrtc.org> Reviewed-by: Erik Språng <sprang@webrtc.org> Commit-Queue: Niels Moller <nisse@webrtc.org> Cr-Commit-Position: refs/heads/master@{#26213}
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.