platform-external-webrtc

Files

fgalligan@google.com 8f06a8aeb0 Update generated asm offsets scripts.

Libvpx updated the unpack scripts to fix building dependencies.

Roll libvpx 269083:278063
See https://codereview.chromium.org/295313002/
https://codereview.chromium.org/298063002/
https://codereview.chromium.org/305533008/
https://codereview.chromium.org/305703002/
https://codereview.chromium.org/298383003/
https://codereview.chromium.org/302863004/
https://codereview.chromium.org/320923003/
https://codereview.chromium.org/325313007/
https://codereview.chromium.org/346563002/
for the libvpx changes.

See https://codereview.chromium.org/313243004/
for the WebView changes.

*NOTE* This CL will break the Android bots as they are built in a
Chromium checkout, which will pull in old libvpx DEPS. They will
cycle to green when we roll libvpx into Chromium. We must do the
rolls in this order because we have to land webrtc and libvpx at
the same time into Chromium.

BUG=377062
TBR=andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/12789004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6481 4adac7df-926f-26a2-2b94-8c16560cd09d

2014-06-18 17:38:08 +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

Update generated asm offsets scripts.

2014-06-18 17:38:08 +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.