webrtc_m130

History

pbos@webrtc.org 919f87fb36 Delete capturers after destroying streams in test.

Since the renderers in CallTest.SendsAndReceiveStreams also stopped the
capturers they must be deleted after the VideoReceiveStream is stopped
or an use-after-free may occur.

R=mflodman@webrtc.org
BUG=

Review URL: https://webrtc-codereview.appspot.com/5969004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5300 4adac7df-926f-26a2-2b94-8c16560cd09d

2013-12-16 14:55:54 +00:00

build

Move realtime tests to webrtc_perf_tests.

2013-12-13 12:48:05 +00:00

common_audio

Roll chromium_revision 232627:238260

2013-12-11 13:37:12 +00:00

common_video

Add SwapFrame() to VideoSendStreamInput.

2013-12-11 16:26:16 +00:00

examples/android

Complete rewrite of demo application.

2013-12-12 16:55:37 +00:00

modules

Revert 5285 "Revert 5228 "Use the RTT from RtcpRttStats class if..."

2013-12-16 14:40:36 +00:00

system_wrappers

Revert r5294 to re-roll r5293.

2013-12-16 12:24:44 +00:00

test

Update talk to 58127566 together with

2013-12-13 00:21:03 +00:00

tools

Roll chromium_revision 232627:238260

2013-12-11 13:37:12 +00:00

video

Delete capturers after destroying streams in test.

2013-12-16 14:55:54 +00:00

video_engine

Revert r5294 to re-roll r5293.

2013-12-16 12:24:44 +00:00

voice_engine

Disabled tests on Android. The issue 2723 is filed to investigate the reason for tests failing.

2013-12-13 22:04:18 +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

Revert r5294 to re-roll r5293.

2013-12-16 12:24:44 +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

Correctly define OVERRIDE when building with g++ 4.7 and C++11 support

2013-12-10 19:20:46 +00:00

video_engine_tests.isolate

Merge metrics_unittests into video_engine_tests.

2013-12-13 14:31:47 +00:00

video_receive_stream.h

Adding REMB to receive stream configuration, the send side will always

2013-12-13 16:36:28 +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

Add SwapFrame() to VideoSendStreamInput.

2013-12-11 16:26:16 +00:00

webrtc_examples.gyp

Complete rewrite of demo application.

2013-12-12 16:55:37 +00:00

webrtc_perf_tests.isolate

Move realtime tests to webrtc_perf_tests.

2013-12-13 12:48:05 +00:00

webrtc_tests.gypi

Revert r5294 to re-roll r5293.

2013-12-16 12:24:44 +00:00

webrtc.gyp

Add SwapFrame() to VideoSendStreamInput.

2013-12-11 16:26:16 +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.