webrtc_m130

History

pbos@webrtc.org 0f3d0bb601 Stop video capturers in multi-stream test.

Expected to reduce runtime and flakiness in
CallTest.SendsAndReceivesMultipleStreams on linux_memcheck which is
presumed to be due to contention between the threads.

BUG=
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/5249004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5238 4adac7df-926f-26a2-2b94-8c16560cd09d

2013-12-06 15:48:17 +00:00

build

Remove the long disabled WEBRTC_SVNREVISION define.

2013-12-04 17:00:44 +00:00

common_audio

Upgrade scoped_ptr to Chromium's latest version.

2013-10-22 12:50:00 +00:00

common_video

Revert "Remove extra copy in VideoCaptureImpl::IncomingFrameI420"

2013-10-31 23:41:04 +00:00

modules

Fix use of uninitialized memory in RtpSenderTest::StreamDataCountersCallbacks

2013-12-06 13:16:44 +00:00

system_wrappers

Move implementation files out of the webrtc/ root.

2013-12-05 12:11:47 +00:00

test

Make VideoSendStreamTest::MaxPacketSize test a whole range of frame sizes, to make sure all corner cases are covered.

2013-12-05 16:15:11 +00:00

tools

Made video quality toolchain more configurable.

2013-11-25 14:10:20 +00:00

video

Stop video capturers in multi-stream test.

2013-12-06 15:48:17 +00:00

video_engine

Fraction lost statistics not being reported

2013-12-06 11:56:55 +00:00

voice_engine

Remove the long disabled WEBRTC_SVNREVISION define.

2013-12-04 17:00:44 +00:00

.gitignore

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

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

call.h

Adds support for sending redundant payloads over RTX.

2013-12-04 10:24:26 +00:00

common_types.h

Add callbacks for send channel rtp statistics

2013-12-05 14:29:02 +00:00

common.h

Add a Config class interface to AudioProcessing for passing options.

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

config.h

Rename RTP-extension constants.

2013-11-20 11:48:56 +00:00

engine_configurations.h

Adds a new voice engine warning for the typing noise off state.

2013-09-17 18:09:20 +00:00

experiments.h

Adds support for sending redundant payloads over RTX.

2013-12-04 10:24:26 +00:00

frame_callback.h

Implement and test EncodedImageCallback in new ViE API.

2013-11-26 11:41:59 +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

PATENTS

Move src/ -> webrtc/

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

PRESUBMIT.py

Modified the presubmit checks such that difference license templates are checked for in webrtc and talk folder.

2013-07-22 22:32:50 +00:00

README.chromium

Move src/ -> webrtc/

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

transport.h

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

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

typedefs.h

Check if WARN_UNUSED_RESULT and COMPILE_ASSERT are defined.

2013-10-23 19:11:32 +00:00

video_engine_tests.isolate

Remove CallTest dependency on voice_engine/test/.

2013-12-05 14:24:17 +00:00

video_receive_stream.h

Set local SSRC for VideoReceiveStream.

2013-12-03 10:13:04 +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

Implement and test EncodedImageCallback in new ViE API.

2013-11-26 11:41:59 +00:00

webrtc_tests.gypi

Remove CallTest dependency on voice_engine/test/.

2013-12-05 14:24:17 +00:00

webrtc.gyp

Move implementation files out of the webrtc/ root.

2013-12-05 12:11:47 +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.