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Sami Kalliomaki 9c0c75bd6e Add GN targets for AppRTC Demo on Android. 
Adds GN equivalents for following targets:
AppRTCDemo          -> //webrtc/examples:AppRTCDemo
AppRTCDemo_apk      -> //webrtc/examples:AppRTCDemo_lib (kind of)
AppRTCDemoJUnitTest -> //webrtc/examples:AppRTCDemoJUnitTest
AppRTCDemoTest      -> //webrtc/examples:AppRTCDemoTest
libjingle_peerconnection_java -> //webrtc/api/libjingle_peerconnection_java
libjingle_peerconnection_so   -> //webrtc/api/libjingle_peerconnection_so

New GN targets:
//webrtc/base:base_java
//webrtc/examples:AppRTCDemo_resources
//webrtc/examples/androidapp/third_party/autobanh:autobanh_java

BUG=webrtc:6035
R=magjed@webrtc.org, tommi@webrtc.org

Review URL: https://codereview.webrtc.org/2105803002 .

Cr-Commit-Position: refs/heads/master@{#13322}
2016-06-29 12:55:12 +00:00
..
androidjunit
Reland of Android: Change camera fps range selection (patchset #1 id:1 of https://codereview.webrtc.org/2021233002/ )
2016-06-01 07:44:07 +00:00
api
Add GN targets for AppRTC Demo on Android.
2016-06-29 12:55:12 +00:00
audio
Move RtcEventLog object from inside VoiceEngine to Call.
2016-06-29 11:57:01 +00:00
base
Add GN targets for AppRTC Demo on Android.
2016-06-29 12:55:12 +00:00
build
Remove dead GYP target audio_device_module_java
2016-06-20 08:43:42 +00:00
call
Move RtcEventLog object from inside VoiceEngine to Call.
2016-06-29 11:57:01 +00:00
common_audio
Revert "Pull out the PostFilter to its own NonlinearBeamformer API"
2016-06-28 17:38:41 +00:00
common_video
Fixing race condition in IncomingVideoStream.
2016-06-28 17:46:44 +00:00
examples
Add GN targets for AppRTC Demo on Android.
2016-06-29 12:55:12 +00:00
libjingle
GN: Update xmpp and p2p to match Chromium build
2016-06-18 11:23:08 +00:00
media
Move RtcEventLog object from inside VoiceEngine to Call.
2016-06-29 11:57:01 +00:00
modules
Move RtcEventLog object from inside VoiceEngine to Call.
2016-06-29 11:57:01 +00:00
p2p
Fix the turn and udp port type.
2016-06-28 16:44:55 +00:00
pc
Move RtcEventLog object from inside VoiceEngine to Call.
2016-06-29 11:57:01 +00:00
sdk
Revert of Delete method cricket::VideoFrame::Copy. (patchset #7 id:120001 of https://codereview.webrtc.org/2080253002/ )
2016-06-21 21:03:01 +00:00
system_wrappers
FileWrapper[Impl] modifications and actually remove the "Impl" class.
2016-06-15 17:30:18 +00:00
test
Move RtcEventLog object from inside VoiceEngine to Call.
2016-06-29 11:57:01 +00:00
tools
New rtc dump analyzing tool in Python
2016-06-20 14:39:21 +00:00
video
Create a null decoder for h264 when not supported, instead of crashing.
2016-06-28 13:39:52 +00:00
voice_engine
Move RtcEventLog object from inside VoiceEngine to Call.
2016-06-29 11:57:01 +00:00
.gitignore
Flesh out webrtc/.gitignore
2015-11-16 19:02:02 +00:00
audio_receive_stream.h
Add AudioReceiveStream::SetGain() method and use that in WVoMC::SetOutputVolume().
2016-06-17 15:30:58 +00:00
audio_send_stream.h
Add AudioSendStream::SetMuted() method and use it in WVoMC::MuteStream().
2016-06-16 17:53:28 +00:00
audio_sink.h
Reland "Move webrtc/audio/audio_sink.h to webrtc/ and fix some dependencies."
2016-02-26 21:46:22 +00:00
audio_state.h
Add comments about the Audio parts of the public Call API being WIP.
2015-12-03 12:06:31 +00:00
BUILD.gn
Build webrtc_nonparallel_tests under GN.
2016-06-27 17:31:56 +00:00
call.h
Move RtcEventLog object from inside VoiceEngine to Call.
2016-06-29 11:57:01 +00:00
codereview.settings
Give a more specific URL for creating WebRTC checkout
2016-04-01 07:36:53 +00:00
common_types.cc
FileWrapper[Impl] modifications and actually remove the "Impl" class.
2016-06-15 17:30:18 +00:00
common_types.h
Avoid unnecessary HW video encoder reconfiguration
2016-06-16 19:08:11 +00:00
common.gyp
Revert of Add missing headers and fix some missing dependencies (patchset #1 id:20001 of https://codereview.webrtc.org/1990593002/ )
2016-05-18 12:00:55 +00:00
common.h
#include "webrtc/base/constructormagic.h" where appropriate
2016-04-26 15:14:48 +00:00
config.cc
Avoid unnecessary HW video encoder reconfiguration
2016-06-16 19:08:11 +00:00
config.h
Avoid unnecessary HW video encoder reconfiguration
2016-06-16 19:08:11 +00:00
DEPS
Moved creation of AudioDecoderFactory to inside PeerConnectionFactory.
2016-06-13 14:35:01 +00:00
engine_configurations.h
Remove the VoEDtmf interface.
2016-03-08 12:11:00 +00:00
LICENSE
…
LICENSE_THIRD_PARTY
Add the Ooura FFT to RealFourier.
2015-03-19 20:07:43 +00:00
OWNERS
Add kwiberg@webrtc.org as root owner.
2016-06-14 09:39:40 +00:00
PATENTS
…
PRESUBMIT.py
…
README.chromium
…
rtc_unittests_apk.isolate
Add isolate files for Android tests
2016-04-18 03:08:28 +00:00
rtc_unittests.isolate
…
supplement.gypi
Cleanup MIPS specific link configuration
2016-06-20 05:39:25 +00:00
transport.h
Reland of Merge webrtc/video_engine/ into webrtc/video/ (patchset #2 id:300001 of https://codereview.webrtc.org/1507903005/ )
2015-12-09 11:13:40 +00:00
typedefs.h
[MIPS] Fix build issue for mips64el
2016-06-23 10:52:35 +00:00
video_decoder.h
Replace scoped_ptr with unique_ptr in webrtc/common_video/
2016-03-02 11:41:40 +00:00
video_encoder.h
Remove QualityScaler framerate reduction.
2016-04-26 11:37:20 +00:00
video_engine_tests_apk.isolate
Add isolate files for Android tests
2016-04-18 03:08:28 +00:00
video_engine_tests.isolate
Add audio streams to CallTest and a first A/V call test.
2016-01-07 16:43:31 +00:00
video_frame.h
Add TODO comments on deprecated VideoFrame methods.
2016-06-22 15:47:52 +00:00
video_receive_stream.h
Movable support for VideoReceiveStream::Config and avoid copies.
2016-06-10 15:58:12 +00:00
video_send_stream.h
Remove webrtc/stream.h and unutilized inheritance.
2016-05-02 03:18:36 +00:00
webrtc_examples.gyp
Add GN targets for AppRTC Demo on Android.
2016-06-29 12:55:12 +00:00
webrtc_nonparallel_tests_apk.isolate
Add isolate files for Android tests
2016-04-18 03:08:28 +00:00
webrtc_nonparallel_tests.isolate
Create isolate files for nonparallel tests.
2015-10-29 14:29:15 +00:00
webrtc_perf_tests_apk.isolate
Add isolate files for Android tests
2016-04-18 03:08:28 +00:00
webrtc_perf_tests.isolate
Fix missing resource file in webrtc_perf_tests.isolate
2016-06-17 07:39:49 +00:00
webrtc_tests.gypi
Implement timestamp translation/filter in VideoCapturer.
2016-06-22 15:36:58 +00:00
webrtc.gyp
Remove webrtc_all target
2016-06-14 09:09:24 +00:00

README.chromium 

Name: WebRTC
URL: http://www.webrtc.org
Version: 90
License: BSD
License File: LICENSE

Description:
WebRTC provides real time voice and video processing
functionality to enable the implementation of 
PeerConnection/MediaStream.

Third party code used in this project is described 
in the file LICENSE_THIRD_PARTY.