This reverts commit 0c15c5332fea2bbf5fe29dd806f9f4e606eeb9b8.
Reason for revert: This causes problems in this moment. I have to fix a target in rtc_base before landing this presubmit check.
Original change's description:
> Adding PRESUBMIT check to avoid mixing C, C++ and Objc-C/Obj-C++.
>
> The error message will be something like:
>
> GN targets cannot mix .c (or .cc) and .m (or .mm) source files.
> Please create a separate target for each collection of sources.
> Mixed sources:
> {
> BUILD_GN_PATH: [
> [
> TARGET_NAME,
> [
> SOURCES
> ]
> ],
> ...
> ],
> ...
> }
>
> Bug: webrtc:7743
> Change-Id: I45dd2c621b830e5aeb081fa8d17c9497a49c2554
> Reviewed-on: https://webrtc-review.googlesource.com/1980
> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
> Reviewed-by: Henrik Kjellander <kjellander@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#19897}
TBR=kjellander@webrtc.org,mbonadei@webrtc.org
Change-Id: I73ff609b0140719473afd36ead1632e5cc3b41f6
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:7743
Reviewed-on: https://webrtc-review.googlesource.com/2180
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#19898}
Revert of Completed the functionalities of SrtpTransport. (patchset 7 id:320001 of https://codereview.webrtc.org/2997983002/ )
Revert of Completed the functionalities of SrtpTransport. (patchset 7 id:320001 of https://codereview.webrtc.org/2997983002/ )
Revert of Fix the video buffer size should take rtt into consideration (patchset #3 id:40001 of https://codereview.chromium.org/2980413002/ )
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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