webrtc

History

andrew@webrtc.org 5804936052 Add format members to AudioConverter for DCHECKing.

And use a std::min. Post-commit fixes after:
https://review.webrtc.org/30779004/

TBR=kwiberg

Review URL: https://webrtc-codereview.appspot.com/25059004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7600 4adac7df-926f-26a2-2b94-8c16560cd09d

2014-11-03 21:32:14 +00:00

base

Only configure the SSL library in one place.

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

build

Add VP9 codec to VCM and vie_auto_test.

2014-11-01 06:10:48 +00:00

common_audio

Add format members to AudioConverter for DCHECKing.

2014-11-03 21:32:14 +00:00

common_video

Update all .isolate files for the new format.

2014-10-31 18:08:09 +00:00

examples

Update Android projects to API level 21.

2014-10-31 23:26:10 +00:00

libjingle

move xmpp and p2p to webrtc

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

modules

Update rate control parameter in vp9 test.

2014-11-03 21:26:08 +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

Add stats for video:

2014-11-03 14:40:38 +00:00

test

Roll chromium_revision: 28d1981..d3db2ff

2014-11-03 20:10:26 +00:00

tools

Update all .isolate files for the new format.

2014-10-31 18:08:09 +00:00

video

Add VP9 codec to VCM and vie_auto_test.

2014-11-01 06:10:48 +00:00

video_engine

Roll chromium_revision: 28d1981..d3db2ff

2014-11-03 20:10:26 +00:00

voice_engine

Update all .isolate files for the new format.

2014-10-31 18:08:09 +00:00

.gitignore

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

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

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 VP9 codec to VCM and vie_auto_test.

2014-11-01 06:10:48 +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

Add VP9 codec to VCM and vie_auto_test.

2014-11-01 06:10:48 +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

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

GN: Add BUILD.gn files + kjellander to OWNERS

2014-06-23 19:21:07 +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

rtc_unittests.isolate

Update all .isolate files for the new format.

2014-10-31 18:08:09 +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

Add VP9 codec to VCM and vie_auto_test.

2014-11-01 06:10:48 +00:00

video_engine_tests.isolate

Update all .isolate files for the new format.

2014-10-31 18:08:09 +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

Update all .isolate files for the new format.

2014-10-31 18:08:09 +00:00

webrtc_tests.gypi

Update all .isolate files for the new format.

2014-10-31 18:08:09 +00:00

webrtc.gyp

Add VP9 codec to VCM and vie_auto_test.

2014-11-01 06:10:48 +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.