platform-external-webrtc

Files

bjornv@webrtc.org cb0ea43e57 audio_processing: Forces extended filter to be used in splitting filter test.

The behavior differ between "normal" and "extended" modes when using AEC. In the extended filter mode nothing is processed until we have received a farend frame. This is exactly what is needed in this part of the splitting filter test.
On Android, we do not use the normal mode, which made the test to fail.

BUG=3445
R=kwiberg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/12679004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6368 4adac7df-926f-26a2-2b94-8c16560cd09d

2014-06-09 08:21:52 +00:00

base

Rebase webrtc/base with r6345 version of talk/base:

2014-06-05 20:40:11 +00:00

build

Make it possible to build webrtc for arm64.

2014-06-04 17:15:42 +00:00

common_audio

The correct fix of workaround in r6261.

2014-06-05 08:53:51 +00:00

common_video

Android: cleanup gtest_target_type conditions.

2014-06-04 20:46:50 +00:00

examples

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

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

modules

audio_processing: Forces extended filter to be used in splitting filter test.

2014-06-09 08:21:52 +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

Android: cleanup gtest_target_type conditions.

2014-06-04 20:46:50 +00:00

test

Make VideoSendStream/VideoReceiveStream configs const.

2014-06-06 10:49:19 +00:00

tools

Android: cleanup gtest_target_type conditions.

2014-06-04 20:46:50 +00:00

video

Make VideoSendStream/VideoReceiveStream configs const.

2014-06-06 10:49:19 +00:00

video_engine

ViEAutoTestAndroid: Unbreak compile by casting void* to jobject.

2014-06-06 18:59:30 +00:00

voice_engine

Fix the chain that propagates the audio frame's rtp and ntp timestamp including:

2014-06-05 20:34:08 +00:00

.gitignore

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

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

call.h

Make VideoSendStream/VideoReceiveStream configs const.

2014-06-06 10:49:19 +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

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

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

Merge metrics_unittests into video_engine_tests.

2013-12-13 14:31:47 +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

Make VideoSendStream/VideoReceiveStream configs const.

2014-06-06 10:49:19 +00:00

webrtc_examples.gyp

Add webrtc field trials API.

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

webrtc_perf_tests.isolate

Move realtime tests to webrtc_perf_tests.

2013-12-13 12:48:05 +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.