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henrik.lundin@webrtc.org ed8b281265 Re-comitting r5711: "Fixing a flaky test in video_engine_tests" 
The CL was reverted in r5712, due to bots going red. However, these bots
are unrelated to this CL.

Original description:
VideoSendStreamTest.SuspendBelowMinBitrate was flaky. The problem was
that when the first non-padding packet was sent after the stream was
resumed, the statistics had not always been updated so that
stats.suspended was false. After seeing the first non-padding packet
after suspension, the test will now go into a state where it waits for
the statistics to be changed.

BUG=3068
R=pbos@webrtc.org
TBR=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/10099004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5713 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-03-18 08:43:29 +00:00
..
build
Enable VS2013 for Windows compilation by default.
2014-03-12 14:41:58 +00:00
common_audio
Add a float interface to PushSincResampler.
2014-03-10 18:51:42 +00:00
common_video
Add an AlignedFreeDeleter and remove scoped_ptr_malloc.
2014-02-20 21:08:36 +00:00
examples
Android, WebRTCDemo: fixes crash issue when pressing switch camera button on devices with only one camera.
2014-01-24 18:42:12 +00:00
modules
Small refactor on send_side_bandwidth_estimation.
2014-03-17 17:07:48 +00:00
system_wrappers
Remove std:: prefixes from C functions in webrtc/.
2014-03-07 15:23:34 +00:00
test
Implement minimum transmit bitrate.
2014-03-13 12:52:27 +00:00
tools
Tool to establish a loopback call via apprtc turn server.
2014-03-17 14:23:22 +00:00
video
Re-comitting r5711: "Fixing a flaky test in video_engine_tests"
2014-03-18 08:43:29 +00:00
video_engine
Implement minimum transmit bitrate.
2014-03-13 12:52:27 +00:00
voice_engine
Voice Engine GetRemoteCSRCs should return the CSRCs from rtp_receiver_ instead of _rtpRtcpModule now.
2014-03-11 16:19:56 +00:00
.gitignore
.gitignore: Add *.mk, created as part of ChromiumOS build
2013-01-04 21:25:42 +00:00
call.h
Add configuration for cpu overuse detection to video send stream.
2014-01-31 10:05:07 +00:00
common_types.h
Add RTCP packet type counter (for getting statistics such as sent/received NACK and FIR).
2014-02-19 11:59:02 +00:00
common.h
Add a Config class interface to AudioProcessing for passing options.
2013-07-25 18:28:29 +00:00
config.h
Wire up statistics in video receive stream of new API
2014-02-07 12:06:29 +00:00
engine_configurations.h
Remove external encryption API for VoE.
2014-02-18 11:27:22 +00:00
experiments.h
Missing include in experiments.h
2014-02-25 09:17:43 +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
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
supplement.gypi
Removes script for generating supplement.gypi also adds git ignore for tools/gn.
2014-01-21 15:54:56 +00:00
transport.h
Rename newapi::Transport::SendRTP()->SendRtp().
2013-11-20 12:17:04 +00:00
typedefs.h
Moves WEBRTC_POSIX define from header file to gyp-settings.
2014-03-07 15:30:21 +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
Make VideoReceiveStream::GetStats() const.
2014-02-07 15:32:45 +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
Re-landing "Routing SuspendChange to VideoSendStream::Stats"
2014-03-13 13:31:21 +00:00
webrtc_examples.gyp
Android example apps: fixes issue where useful failure information was suppressed.
2014-01-21 19:03:51 +00:00
webrtc_perf_tests.isolate
Move realtime tests to webrtc_perf_tests.
2013-12-13 12:48:05 +00:00
webrtc_tests.gypi
Adding NetEq performance test to webrtc_perf_tests
2014-01-10 08:24:04 +00:00
webrtc.gyp
Integrate fake_network_pipe into direct_transport.
2013-12-18 20:28:25 +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.