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}
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/.