webrtc_m130

History

henrik.lundin@webrtc.org 7c6e3d188a Moved voe_neteq_stats_unittest to audio_coding_module_unittest

The design of VoeNetEqStatsTest in voice_engine_unittests depended on
being able to inject a factory for the audio coding module into
voice engine. This functionality is now likely going away, which would
make this test fail to compile. Further, the functionality under test
is mostly ACM functionality, wherefore it makes better sense to test it
at ACM level.

BUG=2996
R=henrika@webrtc.org, turaj@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/12029004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5912 4adac7df-926f-26a2-2b94-8c16560cd09d

2014-04-15 17:59:25 +00:00

build

Make everyone an OWNER for .gyp/.gypi add/delete purposes, non-talk/ edition.

2014-04-14 20:08:03 +00:00

common_audio

Make everyone an OWNER for .gyp/.gypi add/delete purposes, non-talk/ edition.

2014-04-14 20:08:03 +00:00

common_video

Propagate capture ntp timestamp from rtp to renderer.

2014-04-15 17:46:33 +00:00

examples

VideoCaptureAndroid: stop referencing ViERenderer

2014-04-08 22:55:07 +00:00

modules

Moved voe_neteq_stats_unittest to audio_coding_module_unittest

2014-04-15 17:59:25 +00:00

system_wrappers

Make everyone an OWNER for .gyp/.gypi add/delete purposes, non-talk/ edition.

2014-04-14 20:08:03 +00:00

test

Make everyone an OWNER for .gyp/.gypi add/delete purposes, non-talk/ edition.

2014-04-14 20:08:03 +00:00

tools

Make everyone an OWNER for .gyp/.gypi add/delete purposes, non-talk/ edition.

2014-04-14 20:08:03 +00:00

video

Make everyone an OWNER for .gyp/.gypi add/delete purposes, non-talk/ edition.

2014-04-14 20:08:03 +00:00

video_engine

Propagate capture ntp timestamp from rtp to renderer.

2014-04-15 17:46:33 +00:00

voice_engine

Moved voe_neteq_stats_unittest to audio_coding_module_unittest

2014-04-15 17:59:25 +00:00

.gitignore

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

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

call.h

Remove internal codecs from VideoSendStream.

2014-03-19 08:43:57 +00:00

common_types.h

Remove usage of webrtc trace in video processing modules.

2014-04-10 11:30:49 +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

Removes VoECallReport sub API as part of a clean-up operation where the goal is to remove unused APIs.

2014-04-14 14:12:50 +00:00

experiments.h

Adding API for setting bandwidth estimation configurations.

2014-03-25 10:37:31 +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

OWNERS

Make everyone an OWNER for .gyp/.gypi add/delete purposes, non-talk/ edition.

2014-04-14 20:08:03 +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

Fix ARM64 detection.

2014-04-01 01:19:08 +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

Simplify pacer interface.

2014-03-19 10:59:52 +00:00

webrtc_examples.gyp

Make WebRTC Android examples build without sourcing envsetup.sh

2014-04-15 08:35:00 +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.