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(this CL is based on the work by Taylor and Steve in https://webrtc-review.googlesource.com/c/src/+/10201) This SetAudioPlayout method lets applications disable audio playout while still processing incoming audio data and generating statistics on the received audio. This may be useful if the application wants to set up media flows as soon as possible, but isn't ready to play audio yet. Currently, native applications don't have any API point to control this, unless they completely implement their own AudioDeviceModule. The SetAudioRecording works in a similar fashion but for the recorded audio. One difference is that calling SetAudioRecording(false) does not keep any audio processing alive. TBR=solenberg Bug: webrtc:7313 Change-Id: I0aa075f6bfef9818f1080f85a8ff7842fb0750aa Reviewed-on: https://webrtc-review.googlesource.com/16180 Reviewed-by: Henrik Andreassson <henrika@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Commit-Queue: Henrik Andreassson <henrika@webrtc.org> Cr-Commit-Position: refs/heads/master@{#20499}
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, generate GN projects with:
--args='target_os="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 jni/.