f693bfae5f
The following APIs on AudioCodingModule are deprecated with this CL: static int NumberOfCodecs(); static int Codec(int, CodecInst*); static int Codec(const char*, CodecInst*, int, size_t); static int Codec(const char*, int, size_t); absl::optional<CodecInst> SendCodec() const; bool RegisterReceiveCodec(int, const SdpAudioFormat&); int RegisterExternalReceiveCodec(int, AudioDecoder*, int, int, const std::string&); int UnregisterReceiveCodec(uint8_t); int32_t ReceiveCodec(CodecInst*); absl::optional<SdpAudioFormat> ReceiveFormat(); As well as this method on RtpRtcp module: int32_t RegisterSendPayload(const CodecInst&); Bug: webrtc:7626 Change-Id: I1230732136f1fe9048cf74afdeab767ca57ac9ce Reviewed-on: https://webrtc-review.googlesource.com/c/113816 Commit-Queue: Fredrik Solenberg <solenberg@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#26025}
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.
Authoritative list of directories that contain the native API header files.
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