platform-external-webrtc

Files

turaj@webrtc.org 12499ff20b Revert 5711 "Fixing a flaky test in video_engine_tests"

> Fixing a flaky test in video_engine_tests
> 
> 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/10069004

TBR=henrik.lundin@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/10089005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5712 4adac7df-926f-26a2-2b94-8c16560cd09d

2014-03-18 00:23:55 +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

Revert 5711 "Fixing a flaky test in video_engine_tests"

2014-03-18 00:23:55 +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.