platform-external-webrtc

Files

bjornv@webrtc.org 84f8ec1f9c Changes to tests and tools in audio_processing.

- Disables ApmTest.EchoCancellationReportsCorrectDelays
This test relys completely on the structure of how reported system delays are handled in AEC. In addition it assumes a fix setup of delay logging buffers. This test should be refactored.

- Adds flag to turn off reported_delay in audioproc
Now it is feasible to turn on and off the use of reported system delays.

BUG=N/A
R=aluebs@webrtc.org, kwiberg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/16749004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6492 4adac7df-926f-26a2-2b94-8c16560cd09d

2014-06-19 12:14:33 +00:00

base

base: Renaming + conforming: post commit review changes for https://webrtc-codereview.appspot.com/17699005/

2014-06-17 14:48:44 +00:00

build

base: Renaming + conforming: post commit review changes for https://webrtc-codereview.appspot.com/17699005/

2014-06-17 14:48:44 +00:00

common_audio

common_audio: Removes macro WEBRTC_SPL_LSHIFT_U16

2014-06-16 10:30:14 +00:00

common_video

Pass GYP DEPTH variable to isolate.

2014-06-13 09:02:15 +00:00

examples

AppRTCDemo(android): support app (UI) & capture rotation.

2014-06-06 18:40:44 +00:00

modules

Changes to tests and tools in audio_processing.

2014-06-19 12:14:33 +00:00

overrides/webrtc/base

Rename webrtc/base's IS_ALIGNED macro to RTC_IS_ALIGNED to avoid conflict between webrtc/base/basictypes.h and third_party/.../vpx_codec.h.

2014-05-21 16:52:14 +00:00

system_wrappers

Pass GYP DEPTH variable to isolate.

2014-06-13 09:02:15 +00:00

test

Updated W3C getusermedia tests to the latest version of the spec.

2014-06-17 08:46:58 +00:00

tools

Pass GYP DEPTH variable to isolate.

2014-06-13 09:02:15 +00:00

video

Implements start bitrate for new video API.

2014-06-16 08:57:39 +00:00

video_engine

Add max limit of number for overuses. When limit is reached always apply the rampup delay.

2014-06-16 14:27:19 +00:00

voice_engine

Pass GYP DEPTH variable to isolate.

2014-06-13 09:02:15 +00:00

.gitignore

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

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

BUILD.gn

Initial GN work for WebRTC

2014-06-17 08:54:03 +00:00

call.h

Implements start bitrate for new video API.

2014-06-16 08:57:39 +00:00

common_types.h

Add ToString() to VideoSendStream::Config.

2014-05-15 09:35:06 +00:00

common.gyp

Add ToString() to VideoSendStream::Config.

2014-05-15 09:35:06 +00:00

common.h

Add a Config class interface to AudioProcessing for passing options.

2013-07-25 18:28:29 +00:00

config.cc

Add ToString() to VideoSendStream::Config.

2014-05-15 09:35:06 +00:00

config.h

Add ToString() to VideoSendStream::Config.

2014-05-15 09:35:06 +00:00

engine_configurations.h

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

2014-05-12 12:19:19 +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

Add kjellander@webrtc.org as OWNER for *.isolate

2014-06-10 05:42:53 +00:00

PATENTS

Move src/ -> webrtc/

2012-10-22 18:19:23 +00:00

PRESUBMIT.py

PRESUBMIT.py: accept variants on the copyright message that are present in the codebase.

2014-05-23 17:27:18 +00:00

README.chromium

Move src/ -> webrtc/

2012-10-22 18:19:23 +00:00

supplement.gypi

Roll chromium_revision 260462:266514

2014-04-29 09:36:40 +00:00

transport.h

Rename newapi::Transport::SendRTP()->SendRtp().

2013-11-20 12:17:04 +00:00

typedefs.h

Remove ALLOW_UNUSED.

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

video_engine_tests.isolate

Pass GYP DEPTH variable to isolate.

2014-06-13 09:02:15 +00:00

video_receive_stream.h

Rename Start/Stop in Video{Send,Receive}Streams.

2014-04-24 11:13:21 +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

Enable pacing by default and remove the option to disable it from the new API.

2014-06-12 15:12:25 +00:00

webrtc_examples.gyp

Add webrtc field trials API.

2014-05-14 12:24:04 +00:00

webrtc_perf_tests.isolate

Pass GYP DEPTH variable to isolate.

2014-06-13 09:02:15 +00:00

webrtc_tests.gypi

Rename neteq4 folder to neteq

2014-06-09 08:10:28 +00:00

webrtc.gyp

Add ToString() to VideoSendStream::Config.

2014-05-15 09:35:06 +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.