This reverts commit 5117b047875970cf61f2403b590c44c37bfa8272. Reason for revert: Still breaks downstream projects that include too much stuff. Original change's description: > Reland "Clean up libjingle API dependencies." > > This is a reland of 57fb3154b5411934b80051ad827db4e54d00f381 > Original change's description: > > Clean up libjingle API dependencies. > > > > This CL moves candidate.h into the public API, since it has > > been implicitly included before. > > > > This is a straightforward way of solving the circular > > dependencies involving that file. For instance, > > libjingle_peerconnection_api includes candidate.h from > > jsepicecandidate.h, but _api can't depend on rtc_p2p, which > > depends on _api. In fact, _api can't depend on much at all > > since it's a very high level abstraction; instead, things > > should depend on it. > > > > Furthermore, we have the case where deprecated headers > > include headers in internal modules. I just have to turn > > off include checking for those, but that's not a big deal. > > > > This CL punts the problem of callfactoryinterface.h being > > implicitly included, and pulling in most of the call > > module with it. This should be addressed in a follow-up > > CL. > > > > Bug: webrtc:7504 > > Change-Id: I1b1729408158418333ccdf702bf529386090f0d7 > > Reviewed-on: https://webrtc-review.googlesource.com/2020 > > Commit-Queue: Patrik Höglund <phoglund@webrtc.org> > > Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org> > > Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org> > > Cr-Commit-Position: refs/heads/master@{#20034} > > Bug: webrtc:7504 > Change-Id: I74aeeff678a4ce6482d2f402493ae13e698f1390 > Reviewed-on: https://webrtc-review.googlesource.com/4703 > Commit-Queue: Patrik Höglund <phoglund@webrtc.org> > Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org> > Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org> > Reviewed-by: Patrik Höglund <phoglund@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#20062} TBR=phoglund@webrtc.org,deadbeef@webrtc.org,solenberg@webrtc.org Change-Id: I19068df5f3ee8145c5ff13c86a42b6860e9cc834 No-Presubmit: true No-Tree-Checks: true No-Try: true Bug: webrtc:7504 Reviewed-on: https://webrtc-review.googlesource.com/5460 Reviewed-by: Patrik Höglund <phoglund@webrtc.org> Commit-Queue: Patrik Höglund <phoglund@webrtc.org> Cr-Commit-Position: refs/heads/master@{#20065}
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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