8f4bc41c42
This reverts commit ac2f3d14e45398930bc35ff05ed7a3b9b617d328. Reason for revert: Breaks downstream project Original change's description: > Move CryptoOptions to api/crypto from rtc_base/sslstreamadapter.h > > Promotes rtc::CryptoOptions to webrtc::CryptoOptions converting it from class > that only handles SRTP configuration to a more generic structure that can be > used and extended for all per peer connection CryptoOptions that can be on a > given PeerConnection. > > Now all SRTP related options are under webrtc::CryptoOptions::Srtp and can be > accessed as crypto_options.srtp.whatever_option_name. This is more inline with > other structures we have in WebRTC such as VideoConfig. As additional features > are added over time this will allow the structure to remain compartmentalized > and concerned components can only request a subset of the overall configuration > structure e.g: > > void MySrtpFunction(const webrtc::CryptoOptions::Srtp& srtp_config); > > In addition to this it made little sense for sslstreamadapter.h to hold all > Srtp related configuration options. The header has become loo large and takes on > too many responsibilities and spilting this up will lead to more maintainable > code going forward. > > This will be used in a future CL to enable configuration options for the newly > supported Frame Crypto. > > Change-Id: I99d1be36740c59548c8e62db52d68d738649707f > Bug: webrtc:9681 > Reviewed-on: https://webrtc-review.googlesource.com/c/105180 > Reviewed-by: Emad Omara <emadomara@webrtc.org> > Reviewed-by: Kári Helgason <kthelgason@webrtc.org> > Reviewed-by: Sami Kalliomäki <sakal@webrtc.org> > Reviewed-by: Qingsi Wang <qingsi@webrtc.org> > Reviewed-by: Steve Anton <steveanton@webrtc.org> > Commit-Queue: Benjamin Wright <benwright@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#25130} TBR=steveanton@webrtc.org,sakal@webrtc.org,kthelgason@webrtc.org,emadomara@webrtc.org,qingsi@webrtc.org,benwright@webrtc.org Bug: webrtc:9681 Change-Id: Ib0075c477c951b540d4deecb3b0cf8cf86ba0fff Reviewed-on: https://webrtc-review.googlesource.com/c/105541 Reviewed-by: Oleh Prypin <oprypin@webrtc.org> Commit-Queue: Oleh Prypin <oprypin@webrtc.org> Cr-Commit-Position: refs/heads/master@{#25133}
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