This reverts commit 2ff7ecfceb99deb3f2351c4184f84c1322568087. Reason for revert: <INSERT REASONING HERE> Original change's description: > Reduce max possible size of map that holds encoded frame info. > > Bug: webrtc:8375 > Change-Id: Idc57e68dc44fd73e5c0aa85d82c1e3659d8ea292 > Reviewed-on: https://webrtc-review.googlesource.com/8301 > Reviewed-by: Rasmus Brandt <brandtr@webrtc.org> > Reviewed-by: Erik Språng <sprang@webrtc.org> > Commit-Queue: Åsa Persson <asapersson@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#20232} TBR=brandtr@webrtc.org,asapersson@webrtc.org,sprang@webrtc.org Change-Id: I1dcb7ab588e5ab3eb79bec2c39f615480e31c3bb No-Presubmit: true No-Tree-Checks: true No-Try: true Bug: webrtc:8375 Reviewed-on: https://webrtc-review.googlesource.com/8460 Reviewed-by: Åsa Persson <asapersson@webrtc.org> Commit-Queue: Åsa Persson <asapersson@webrtc.org> Cr-Commit-Position: refs/heads/master@{#20247}
Reland of Fix the video buffer size should take rtt into consideration (patchset #2 id:160001 of https://codereview.chromium.org/3002033002/ )
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 http://www.webrtc.org/native-code/development for instructions on how to get started developing with the native code.
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: http://build.chromium.org/p/client.webrtc
- Coding style guide
- Code of conduct
Description
The idea is to make CMake build for WebRTC m130 version - for audio processing module
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