webrtc_m130

History

henrik.lundin@webrtc.org fcfc6a990e Small refactoring of NetEq unittest for CNG with clock drift

Converting the test to a method within the test fixture, and setting
up two tests that call this method. One for positive and one for
negative clock drift. The one with positive clock drift is disabled
for now since it does not pass, but will be re-enabled shortly.

This change is only made for NetEq4.

R=tlegrand@google.com

Review URL: https://webrtc-codereview.appspot.com/8599004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5541 4adac7df-926f-26a2-2b94-8c16560cd09d

2014-02-13 11:42:28 +00:00

build

Rename merged webrtc lib to libwebrtc_merged.a.

2014-02-11 18:22:29 +00:00

common_audio

Remove the requirement to call set_sample_rate_hz and friends.

2014-01-07 17:45:09 +00:00

common_video

Add SwapFrame() to VideoSendStreamInput.

2013-12-11 16:26:16 +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 refactoring of NetEq unittest for CNG with clock drift

2014-02-13 11:42:28 +00:00

system_wrappers

Use system's cpu_features library

2014-02-05 18:43:46 +00:00

test

Wire up RTX in VideoReceiveStream.

2014-01-24 09:30:53 +00:00

tools

Now printing less output from compare_videos.py.

2014-01-07 17:59:30 +00:00

video

Make VideoReceiveStream::GetStats() const.

2014-02-07 15:32:45 +00:00

video_engine

Set pacing bitrates in SetEncoder.

2014-02-12 14:50:29 +00:00

voice_engine

Remove unused and not working voe_extended_test.

2014-02-12 09:58:49 +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

Wire up statistics in video receive stream of new API

2014-02-07 12:06:29 +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 ViE external encryption API.

2014-02-11 15:27:49 +00:00

experiments.h

Add experiment: SkipEncodingUnusedStreams

2014-02-10 09:20:51 +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

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

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

Wire up statistics in video receive stream of new API

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