platform-external-webrtc

Files

bjornv@webrtc.org b51d3ea593 Revert 6341 "Fixes and enables SystemDelayTests."

> Fixes and enables SystemDelayTests.
> 
> The root cause for failure was that the delay handling of reported delays was bypassed on Android, whereas the tests assumes that part of AEC to be run.
> This CL checks if it is in use.
> 
> BUG=3445
> R=kwiberg@webrtc.org
> 
> Review URL: https://webrtc-codereview.appspot.com/12689005

TBR=bjornv@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/13589004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6343 4adac7df-926f-26a2-2b94-8c16560cd09d

2014-06-05 13:41:33 +00:00

base

Rebase webrtc/base with r6250:

2014-05-29 15:53:39 +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

WebRTCDemo: clean the error message due to API clean up and add ability to route the audio through all three outputs, headset/earpiece/loudspeaker

2014-05-21 03:37:45 +00:00

modules

Revert 6341 "Fixes and enables SystemDelayTests."

2014-06-05 13:41:33 +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

Android: cleanup gtest_target_type conditions.

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

test

Adding back platform specific renderer to video loopback test.

2014-06-05 09:32:51 +00:00

tools

Android: cleanup gtest_target_type conditions.

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

video

Adding back platform specific renderer to video loopback test.

2014-06-05 09:32:51 +00:00

video_engine

Have RTX be enabled by setting an RTX payload type instead of by setting an RTX SSRC.

2014-06-05 08:25:29 +00:00

voice_engine

Have RTX be enabled by setting an RTX payload type instead of by setting an RTX SSRC.

2014-06-05 08:25:29 +00:00

.gitignore

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

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

call.h

Add DeliveryStatus enum to DeliverPacket().

2014-05-14 13:57:12 +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

Have RTX be enabled by setting an RTX payload type instead of by setting an RTX SSRC.

2014-06-05 08:25:29 +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

Adding back platform specific renderer to video loopback test.

2014-06-05 09:32:51 +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.