14137a1064
Goal is to be able to retrieve more details about possible microphone conflicts in cases where Init/Start of audio recording fails. Only supported on Android N and higher. Also adds new boolean UMA histogram called WebRTC.Audio.SourceMatchesRecordingSession. Its value is stored after the recording session has been stopped. Does not affect the media flow or functionality of the ADM. Time to start audio should not be affected either since the new check and logging takes place on a separate ExecutorService thread. See go/webrtc-adm-android for more details and examples. Bug: webrtc:10971 Change-Id: Ia80c1534e326907a1582824225d5f58caa016922 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/150793 Commit-Queue: Henrik Andreassson <henrika@webrtc.org> Reviewed-by: Alex Glaznev <glaznev@webrtc.org> Reviewed-by: Paulina Hensman <phensman@webrtc.org> Cr-Commit-Position: refs/heads/master@{#29236}
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/.