This reverts commit 3ed32accc2efab456ec4eedf9df4cef1df6b357d. Reason for revert: Compilation failure. Original change's description: > Remove precompiled header for AppRTCMobile. > > Bug: None > Change-Id: Ia46fc3a237a882acef5218ef22c283fb9c379e44 > Reviewed-on: https://webrtc-review.googlesource.com/3340 > Reviewed-by: Anders Carlsson <andersc@webrtc.org> > Commit-Queue: Kári Helgason <kthelgason@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#19948} TBR=andersc@webrtc.org,kthelgason@webrtc.org Change-Id: Id6fc5d4978315be13da7ef03438c0804fa19c4a1 No-Presubmit: true No-Tree-Checks: true No-Try: true Bug: None Reviewed-on: https://webrtc-review.googlesource.com/3500 Reviewed-by: Charu Jain <charujain@webrtc.org> Commit-Queue: Charu Jain <charujain@webrtc.org> Cr-Commit-Position: refs/heads/master@{#19952}
Revert of Add full stack tests for MediaCodec. (patchset #10 id:180001 of https://codereview.webrtc.org/3005253002/ )
Reland of Fix the video buffer size should take rtt into consideration (patchset #2 id:160001 of https://codereview.chromium.org/3002033002/ )
Reland of Fix the video buffer size should take rtt into consideration (patchset #2 id:160001 of https://codereview.chromium.org/3002033002/ )
Revert of Add full stack tests for MediaCodec. (patchset #10 id:180001 of https://codereview.webrtc.org/3005253002/ )
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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