platform-external-webrtc

Files

primiano@chromium.org 5cf73962e6 Fix gyp for video_capture/ensure_initialized.cc.

This is a follow-up to
https://webrtc-codereview.appspot.com/11359004
which introduced an invalid dependency in the
chromium build when building without linker GC.

BUG=2974,3152,chromium:354405
R=andrew@webrtc.org, fischman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/11789005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5898 4adac7df-926f-26a2-2b94-8c16560cd09d

2014-04-14 17:26:31 +00:00

build

Unbreak android APK buildbots by emptying the video_capture_tests_apk target.

2014-04-09 02:34:50 +00:00

common_audio

Add fir_filter to common_audio

2014-03-24 10:16:11 +00:00

common_video

Remove WEBRTC_TRACE use in common_video/

2014-04-07 07:29:18 +00:00

examples

VideoCaptureAndroid: stop referencing ViERenderer

2014-04-08 22:55:07 +00:00

modules

Fix gyp for video_capture/ensure_initialized.cc.

2014-04-14 17:26:31 +00:00

system_wrappers

Remove usage of webrtc trace in video processing modules.

2014-04-10 11:30:49 +00:00

test

VoE changes to allow forwarding of packets from VoE to ViE BWE.

2014-03-24 10:38:25 +00:00

tools

Fix loopback test for case where no constraint is given.

2014-04-10 14:21:45 +00:00

video

Disable UsesTraceCallback

2014-04-10 14:39:22 +00:00

video_engine

Cleaned up logging in video_coding.

2014-04-11 14:08:35 +00:00

voice_engine

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

.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

Remove usage of webrtc trace in video processing modules.

2014-04-10 11:30:49 +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

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

Android example apps: fixes issue where useful failure information was suppressed.

2014-01-21 19:03:51 +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.