platform-external-webrtc

Files

pbos@webrtc.org a6cefcaceb gn: Fix cflags usage

R=brettw@chromium.org
TBR=kjellander@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/29519004

Patch from Cem Kocagil <ckocagil@chromium.org>.

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7198 4adac7df-926f-26a2-2b94-8c16560cd09d

2014-09-16 17:57:02 +00:00

base

Fix GN for rtc_base_approved target.

2014-09-16 11:16:12 +00:00

build

Set minimum SDK level to 10.7 for Mac and iOS.

2014-09-15 08:02:43 +00:00

common_audio

gn: Fix cflags usage

2014-09-16 17:57:02 +00:00

common_video

Revert 7114 "Expose VideoEncoders with webrtc/video_encoder.h."

2014-09-11 09:48:30 +00:00

examples

Fix compile error on JDK 1.7.

2014-09-10 12:35:59 +00:00

libjingle/xmllite

Put base tests in webrtc_tests.gyp

2014-09-10 17:28:19 +00:00

modules

Mark all virtual overrides in the hierarchies of UdpTransportData and

2014-09-16 15:29:02 +00:00

overrides/webrtc/base

Partial implementation of rtc::LogMessage in chromium overrides.

2014-09-15 19:16:21 +00:00

sound

Put base tests in webrtc_tests.gyp

2014-09-10 17:28:19 +00:00

system_wrappers

Add a target for the approved subset of rtc_base.

2014-09-16 01:03:29 +00:00

test

Mark all virtual overrides in the hierarchies of UdpTransportData and

2014-09-16 15:29:02 +00:00

tools

Fix printing of error stack in rtcbot when a test fails via test.fail().

2014-09-10 14:35:35 +00:00

video

Re-enable rampup_tests.cc for Android.

2014-09-15 12:27:35 +00:00

video_engine

Mark all virtual overrides in the hierarchy of RtpData and RtpReceiver as such.

2014-09-12 07:42:33 +00:00

voice_engine

Calculating round-trip-time in send-only channel in VoE.

2014-09-11 07:51:53 +00:00

.gitignore

…

BUILD.gn

Fix the RTC+Chromium GN build.

2014-09-09 19:15:33 +00:00

call.h

Network up/down signaling in Call.

2014-09-03 16:17:12 +00:00

common_types.h

Move pacer to fully use webrtc::Clock instead of webrtc::TickTime.

2014-07-11 13:44:02 +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

Check before send/receive rtp header extensions.

2014-07-20 15:27:35 +00:00

engine_configurations.h

Add boilerplate code for H.264.

2014-07-04 12:42:07 +00:00

experiments.h

Remove no longer used SkipEncodingUnusedStreams.

2014-07-22 07:17:17 +00:00

frame_callback.h

Wire up statistics in video receive stream of new API

2014-02-07 12:06:29 +00:00

LICENSE

…

LICENSE_THIRD_PARTY

Consolidate all third party licenses in LICENSE_THIRD_PARTY.

2013-05-05 18:54:10 +00:00

OWNERS

GN: Add BUILD.gn files + kjellander to OWNERS

2014-06-23 19:21:07 +00:00

PATENTS

…

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

…

supplement.gypi

Roll chromium_revision 289723:291647

2014-08-25 14:16:32 +00:00

transport.h

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

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

typedefs.h

Add CHECK and friends from Chromium.

2014-08-28 16:28:26 +00:00

video_engine_tests.isolate

Pass GYP DEPTH variable to isolate.

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

video_receive_stream.h

Change return value for number of discarded packets to be int.

2014-09-04 07:07:44 +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

Move pacer to fully use webrtc::Clock instead of webrtc::TickTime.

2014-07-11 13:44:02 +00:00

webrtc_examples.gyp

Fix rm command for class cleanup in r7091

2014-09-06 22:11:28 +00:00

webrtc_perf_tests.isolate

Pass GYP DEPTH variable to isolate.

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

webrtc_tests.gypi

Put base tests in webrtc_tests.gyp

2014-09-10 17:28:19 +00:00

webrtc.gyp

Put base tests in webrtc_tests.gyp

2014-09-10 17:28:19 +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.