webrtc/sdk/android
Benjamin Wright a54daf162f Reland "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.

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}
2018-10-11 23:09:07 +00:00
..
api/org/webrtc Add certificate gen/set functionality to bring Android closer to JS API 2018-10-10 13:37:47 +00:00
instrumentationtests Add certificate gen/set functionality to bring Android closer to JS API 2018-10-10 13:37:47 +00:00
native_api Reland "[cleanup] Remove useless includes." 2018-10-08 07:44:19 +00:00
native_unittests Add unit test for JavaToNativeVideoCodecInfo. 2018-07-09 13:26:34 +00:00
src Reland "Move CryptoOptions to api/crypto from rtc_base/sslstreamadapter.h" 2018-10-11 23:09:07 +00:00
tests/src/org/webrtc Add support for platform software video decoder implementations. 2018-09-05 15:15:27 +00:00
AndroidManifest.xml Moving src/webrtc into src/. 2017-09-15 04:25:06 +00:00
BUILD.gn Add certificate gen/set functionality to bring Android closer to JS API 2018-10-10 13:37:47 +00:00
OWNERS Android: Add henrika@ as owner of audio code 2018-03-21 09:59:18 +00:00
PRESUBMIT.py Moving src/webrtc into src/. 2017-09-15 04:25:06 +00:00
README Updating android/README. 2018-03-01 20:22:48 +00:00

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/.