NetEq already uses SdpAudioFormat internally; this CL adds an AudioCodingModule::RegisterReceiveCodec overload that accepts SdpAudioFormat, and propagates it through AcmReceiver into NetEq. The intention is to get rid of the other ways to specify decoders and always use SdpAudioFormat. (And eventually to do the same for encoders too.) NOTRY=true BUG=5801 Review-Url: https://codereview.webrtc.org/2365653004 Cr-Commit-Position: refs/heads/master@{#14506}
Revert of Only expose gflags target in non-Chromium and non-fuzzer builds. (patchset #1 id:40001 of https://codereview.webrtc.org/2321963002/ )
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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. This page is maintained by the Google Chrome team.
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://chromium.googlesource.com/external/webrtc
- 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
Description
The idea is to make CMake build for WebRTC m130 version - for audio processing module
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