Jonas Oreland 358c94f5dd AudioState extensions
This patch modifies AudioState to always call InitRecording
before StartRecording(). This makes it possible to do
SetRecording(false) + SetRecording(true), which before this
patch would not actually work if there was sending streams.

The only way was to add/remove streams...via SDP operations, puh :(.

Bonus: We also needed to modifu AndroidAudioDeviceModule
(which is a thin wrapper) so that StopRecording() will
call AudioInput->StopRecording() even when recording is not
enabled.

BUG=b/397376626

Change-Id: I954b5caab11225b544c3e6a78c5dde357d4eedb5
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/378140
Commit-Queue: Jonas Oreland <jonaso@webrtc.org>
Auto-Submit: Jonas Oreland <jonaso@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43946}
2025-02-20 23:50:12 -08:00
..
2025-02-20 23:50:12 -08:00
2023-11-07 09:58:37 +00:00
2022-03-31 10:48:31 +00:00
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/.