admin/webrtc_m130
1
0
Fork 0
Code Issues Pull Requests Actions Packages Projects Releases Wiki Activity
webrtc_m130/webrtc
History
bjornv@webrtc.org e9d3760d5c AEC: Adds a reported_delay_enabled_ flag 
Adds a feature to completely turn on or off buffer handling based on reported delay values. During startup, reported delays are controlled differently through, e.g., WEBRTC_UNTRUSTED_DELAY. By default, the feature is enabled giving the same output as before this change.

TESTED=trybots, modules_unittest
R=aluebs@webrtc.org, andrew@webrtc.org, kwiberg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/12349005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5965 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-04-23 13:20:07 +00:00
..
build
Set include_internal_video_capture=1 for video_capture_tests
2014-04-16 12:59:49 +00:00
common_audio
common_audio: VADFree() now returns void
2014-04-22 04:45:35 +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
AEC: Adds a reported_delay_enabled_ flag
2014-04-23 13:20:07 +00:00
system_wrappers
Add Chromium's ScopedVector.
2014-04-18 21:20:54 +00:00
test
Remove use of tmpnam.
2014-04-16 08:04:26 +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
Let A/V sync test use default AudioCoding module
2014-04-22 08:36:33 +00:00
video_engine
Propagate capture ntp timestamp from rtp to renderer.
2014-04-15 17:46:33 +00:00
voice_engine
Remove ACM1/ACM2 switching from VoiceEngine tests
2014-04-23 08:39:41 +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
Removes parts of the VoEBase sub API as part of a clean-up operation where the goal is to remove unused APIs.
2014-04-17 10:45:01 +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.