platform-external-webrtc

Files

andrew@webrtc.org 508c91683c Build fix for MIPS32R6.

Exclude MIPS optimizations for MIPS32R6 build since some of the instructions
are not supported. This is temporary fix, until the MIPS32R6 code is added.

R=andrew@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/25989004

Patch from Ljubomir Papuga <lpapuga@mips.com>.

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7580 4adac7df-926f-26a2-2b94-8c16560cd09d

2014-10-31 16:26:17 +00:00

base

Only configure the SSL library in one place.

2014-10-27 18:13:40 +00:00

build

Build fix for MIPS32R6.

2014-10-31 16:26:17 +00:00

common_audio

Build fix for MIPS32R6.

2014-10-31 16:26:17 +00:00

common_video

GN: Add common configs to all targets.

2014-09-28 17:37:22 +00:00

examples

Split video engine android initialization into each internal module initialization.

2014-09-17 11:44:51 +00:00

libjingle

move xmpp and p2p to webrtc

2014-10-28 22:20:11 +00:00

modules

Build fix for MIPS32R6.

2014-10-31 16:26:17 +00:00

overrides

webrtc/overrides: add OWNERS-file.

2014-09-17 08:04:28 +00:00

p2p

move xmpp and p2p to webrtc

2014-10-28 22:20:11 +00:00

sound

rtc_unittest: turned sound's test gyp into gypi to speed up GYP generation.

2014-10-01 16:33:03 +00:00

system_wrappers

Fix a name collision with Android libc++

2014-10-31 16:01:25 +00:00

test

Use external VideoDecoders in VideoReceiveStream.

2014-10-29 15:28:39 +00:00

tools

Adding the subtool rtcBot report visualizer

2014-10-24 09:26:16 +00:00

video

Implement conference-mode temporal-layer screencast.

2014-10-31 13:08:10 +00:00

video_engine

Remove unused code in overuse detector.

2014-10-29 10:05:21 +00:00

voice_engine

Add a simple AudioConverter class.

2014-10-27 18:18:17 +00:00

.gitignore

…

BUILD.gn

arm64 iOS build.

2014-10-31 00:14:39 +00:00

call.h

Remove -1 from Call::Config::start_bitrate_bps.

2014-10-14 11:52:10 +00:00

common_types.h

Add stats for duplicate sent and received NACK requests.

2014-10-29 12:42:30 +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

Implement conference-mode temporal-layer screencast.

2014-10-31 13:08:10 +00:00

config.h

Implement conference-mode temporal-layer screencast.

2014-10-31 13:08:10 +00:00

engine_configurations.h

Enable G.722 for Chromium builds

2014-10-29 08:32:44 +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

…

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

…

rtc_unittests.isolate

Adds isolate for rtc_unittests and moves sound's unittests to rtc_unittest.

2014-09-30 14:21:10 +00:00

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_decoder.h

Use external VideoDecoders in VideoReceiveStream.

2014-10-29 15:28:39 +00:00

video_encoder.h

Revert cls (original cl + fixes) 7422-7424 "Add VP9 codec to VCM..."

2014-10-17 18:54:46 +00:00

video_engine_tests.isolate

Pass GYP DEPTH variable to isolate.

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

video_frame.h

Revert 7151 "Revert 7114 "Expose VideoEncoders with webrtc/video_encoder.h.""

2014-09-17 09:02:25 +00:00

video_receive_stream.h

Configure A/V sync in WebRtcVideoEngine2.

2014-10-31 12:59:34 +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 min transmit bitrate to VideoEncoderConfig.

2014-10-24 09:23:21 +00:00

webrtc_examples.gyp

Add macros and APIs for webrtc histograms.

2014-10-23 12:57:56 +00:00

webrtc_perf_tests.isolate

Pass GYP DEPTH variable to isolate.

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

webrtc_tests.gypi

move xmpp and p2p to webrtc

2014-10-28 22:20:11 +00:00

webrtc.gyp

move xmpp and p2p to webrtc

2014-10-28 22:20:11 +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.