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![]() 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. Reland Fix: - cryptooptions.h - now has enable_aes128_sha1_32_crypto_cipher as an optional root level configuration. - peerconnectionfactory - If this optional is set will now overwrite the underyling value. This along with the other field will be deprecated once dependent projects are updated. TBR=sakal@webrtc.org,kthelgason@webrtc.org,emadomara@webrtc.org,qingsi@webrtc.org Bug: webrtc:9681 Change-Id: Iaa6b741baafb85d352e42f54226119f19d97151d Reviewed-on: https://webrtc-review.googlesource.com/c/105560 Reviewed-by: Benjamin Wright <benwright@webrtc.org> Reviewed-by: Steve Anton <steveanton@webrtc.org> Reviewed-by: Emad Omara <emadomara@webrtc.org> Commit-Queue: Benjamin Wright <benwright@webrtc.org> Cr-Commit-Position: refs/heads/master@{#25135} |
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api/org/webrtc | ||
instrumentationtests | ||
native_api | ||
native_unittests | ||
src | ||
tests/src/org/webrtc | ||
AndroidManifest.xml | ||
BUILD.gn | ||
OWNERS | ||
PRESUBMIT.py | ||
README |
This directory holds a Java implementation of the webrtc::PeerConnection API, as well as the JNI glue C++ code that lets the Java implementation reuse the C++ implementation of the same API. To build the Java API and related tests, make sure you have a WebRTC checkout with Android specific parts. This can be used for linux development as well by configuring gn appropriately, as it is a superset of the webrtc checkout: fetch --nohooks webrtc_android gclient sync You also must generate GN projects with: --args='target_os="android" target_cpu="arm"' More information on getting the code, compiling and running the AppRTCMobile app can be found at: https://webrtc.org/native-code/android/ To use the Java API, start by looking at the public interface of org.webrtc.PeerConnection{,Factory} and the org.webrtc.PeerConnectionTest. To understand the implementation of the API, see the native code in src/jni/pc/.