webrtc_m130

History

Paulina Hensman 24bebb86bd Add checks and offset when using byteBuffer in WebRtcAudioRecord.

See bug for more info.

In this case, the offset of the byteBuffer was observed to be 4 bytes
when testing, meaning that the first 4 bytes sent to the AudioSamples
callback were empty, and the last 4 bytes that should have been sent
were not sent.

This CL adjusts the range copied from the backing array to match the
offset.

Bug: webrtc:9175
Change-Id: I40ac6e10c6d7058ead7eff1c9fa2f342920cf2a4
Reviewed-on: https://webrtc-review.googlesource.com/75123
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Commit-Queue: Paulina Hensman <phensman@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23172}

2018-05-08 12:36:22 +00:00

api/org/webrtc

Adding PeerConnection.Observer.onTrack to the Java SDK.

2018-05-07 18:22:48 +00:00

instrumentationtests

Re-enable PeerConnectionTest#testTrackRemovalAndAddition.

2018-05-08 09:57:26 +00:00

native_api

Increase logging for Java ADM

2018-04-24 13:50:06 +00:00

native_unittests

Add unit tests for Android audio device module

2018-04-12 09:27:45 +00:00

src

Add checks and offset when using byteBuffer in WebRtcAudioRecord.

2018-05-08 12:36:22 +00:00

tests/src/org/webrtc

Add VideoEncoder.ScalingSettings.toString method.

2018-04-18 14:36:19 +00:00

AndroidManifest.xml

Moving src/webrtc into src/.

2017-09-15 04:25:06 +00:00

BUILD.gn

Break out the part of the iSAC codec that's used for Voice Activity Detection

2018-05-04 08:53:34 +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

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/.