webrtc

History

kjellander@webrtc.org a32d15448d Disable tests failing on Android ARM64 (Nexus9).

BUG=4198,4199,4200
TBR=andrew@webrtc.org
TESTED=Printed using #pragma message to check that the define was properly used.

Review URL: https://webrtc-codereview.appspot.com/33919004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@8090 4adac7df-926f-26a2-2b94-8c16560cd09d

2015-01-19 12:46:01 +00:00

base

Re-enable the messagequeue unittests. These were commented out at one point but never reenabled.

2015-01-16 17:52:53 +00:00

build

Fix searching for DirectX SDK during GN build.

2015-01-14 21:25:25 +00:00

common_audio

Add beamforming to audioproc_float utility.

2015-01-15 01:28:36 +00:00

common_video

Move system_wrappers.gyp files to the proper directory.

2015-01-14 09:30:52 +00:00

examples

Use int64_t more consistently for times, in particular for RTT values.

2015-01-12 21:51:21 +00:00

libjingle

Added support for an Origin header in STUN messages.

2015-01-10 00:47:02 +00:00

modules

Disable tests failing on Android ARM64 (Nexus9).

2015-01-19 12:46:01 +00:00

overrides

Remove COMPILE_ASSERT and use static_assert everywhere

2015-01-14 10:51:54 +00:00

p2p

release the turn allocation by sending a refresh request with lifetime 0

2015-01-17 00:58:15 +00:00

sound

rtc_unittest: turned sound's test gyp into gypi to speed up GYP generation.

2014-10-01 16:33:03 +00:00

system_wrappers

Clean unnecessary workaround for chromium import.

2015-01-15 09:12:45 +00:00

test

Move system_wrappers.gyp files to the proper directory.

2015-01-14 09:30:52 +00:00

tools

Move system_wrappers.gyp files to the proper directory.

2015-01-14 09:30:52 +00:00

video

Log configs when creating video streams in Call.

2015-01-15 10:09:39 +00:00

video_engine

Remove unused private data member engine_id_

2015-01-19 07:54:29 +00:00

voice_engine

Remove dual stream functionality in VoiceEngine

2015-01-14 16:07:26 +00:00

.gitignore

.gitignore: Add *.mk, created as part of ChromiumOS build

2013-01-04 21:25:42 +00:00

BUILD.gn

Enabling building with NEON on ARM64

2014-11-26 17:01:40 +00:00

call.h

Use int64_t more consistently for times, in particular for RTT values.

2015-01-12 21:51:21 +00:00

codereview.settings

Add codereview.settings to the /webrtc subdirectory

2014-12-05 13:43:35 +00:00

common_types.h

Refactor some receive-side stats.

2014-12-18 13:50:16 +00:00

common.gyp

Add ToString() to VideoSendStream::Config.

2014-05-15 09:35:06 +00:00

common.h

Add a Config class interface to AudioProcessing for passing options.

2013-07-25 18:28:29 +00:00

config.cc

Revert 8028 "Support associated payload type when registering Rt..."

2015-01-09 20:22:46 +00:00

config.h

Revert 8028 "Support associated payload type when registering Rt..."

2015-01-09 20:22:46 +00:00

engine_configurations.h

Add VP9 codec to VCM and vie_auto_test.

2014-11-01 06:10:48 +00:00

experiments.h

Remove no longer used SkipEncodingUnusedStreams.

2014-07-22 07:17:17 +00:00

frame_callback.h

Wire up statistics in video receive stream of new API

2014-02-07 12:06:29 +00:00

LICENSE

Move src/ -> webrtc/

2012-10-22 18:19:23 +00:00

LICENSE_THIRD_PARTY

Consolidate all third party licenses in LICENSE_THIRD_PARTY.

2013-05-05 18:54:10 +00:00

OWNERS

GN: Add BUILD.gn files + kjellander to OWNERS

2014-06-23 19:21:07 +00:00

PATENTS

Move src/ -> webrtc/

2012-10-22 18:19:23 +00:00

PRESUBMIT.py

PRESUBMIT.py: accept variants on the copyright message that are present in the codebase.

2014-05-23 17:27:18 +00:00

README.chromium

Move src/ -> webrtc/

2012-10-22 18:19:23 +00:00

rtc_unittests.isolate

Update all .isolate files for the new format.

2014-10-31 18:08:09 +00:00

supplement.gypi

Roll chromium_revision 289723:291647

2014-08-25 14:16:32 +00:00

transport.h

Rename newapi::Transport::SendRTP()->SendRtp().

2013-11-20 12:17:04 +00:00

typedefs.h

Annotate COMPILE_ASSERT with __attribute__((unused)).

2014-11-17 13:47:38 +00:00

video_decoder.h

Add support for VP9 in webrtc::Call and video_loopback.

2014-11-04 19:41:15 +00:00

video_encoder.h

Use int64_t more consistently for times, in particular for RTT values.

2015-01-12 21:51:21 +00:00

video_engine_tests.isolate

Update isolate files for Android APK tests.

2014-11-13 08:35:05 +00:00

video_frame.h

Report encoded frame size in VideoSendStream.

2014-12-01 15:23:21 +00:00

video_receive_stream.h

Log configs when creating video streams in Call.

2015-01-15 10:09:39 +00:00

video_renderer.h

Separate Call API/build files from video_engine/.

2013-10-28 16:32:01 +00:00

video_send_stream.h

Log configs when creating video streams in Call.

2015-01-15 10:09:39 +00:00

webrtc_examples.gyp

Move system_wrappers.gyp files to the proper directory.

2015-01-14 09:30:52 +00:00

webrtc_perf_tests.isolate

Update isolate files for Android APK tests.

2014-11-13 08:35:05 +00:00

webrtc_tests.gypi

Move system_wrappers.gyp files to the proper directory.

2015-01-14 09:30:52 +00:00

webrtc.gyp

Move system_wrappers.gyp files to the proper directory.

2015-01-14 09:30:52 +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.