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b732bd5fb5
Add timestamps to the function AudioDeviceBuffer::SetRecordedBuffer. This will be used to store audio timestaps in future changes. This is a part of the A/V sync metric metric feature for mobile. The metric have already launched for web clients. Bug: webrtc:13609 Change-Id: I0031843476ff1b573b262308fca52d587fae30b7 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/249085 Reviewed-by: Henrik Andreassson <henrika@webrtc.org> Reviewed-by: Minyue Li <minyue@google.com> Commit-Queue: Olov Brändström <brandstrom@google.com> Cr-Commit-Position: refs/heads/main@{#35851} |
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| .. | ||
| api/org/webrtc | ||
| instrumentationtests | ||
| native_api | ||
| native_unittests | ||
| src | ||
| tests | ||
| 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/.