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pbos@webrtc.org 3bf3d238c8 Configure A/V sync in WebRtcVideoEngine2. 
Sets up A/V sync for the first video receive channel with the default
voice channel. This is only done when conference mode is disabled to
preserve existing behavior. Ideally we'd know which voice channel to
sync with here.

R=mflodman@webrtc.org, stefan@webrtc.org
BUG=1788

Review URL: https://webrtc-codereview.appspot.com/23249004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@7577 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-10-31 12:59:34 +00:00
..
base
Only configure the SSL library in one place.
2014-10-27 18:13:40 +00:00
build
arm64 iOS build.
2014-10-31 00:14:39 +00:00
common_audio
Revert "Revert part of r7561, "Refactor audio conversion functions.""
2014-10-31 04:58:14 +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
Simplify bwe tests.
2014-10-31 10:47:12 +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
scoped_ptr.h: Renames function and change namespace scope to fix conflicts with Chromium not detected by the FYI bots.
2014-10-28 18:06:42 +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
Configure A/V sync in WebRtcVideoEngine2.
2014-10-31 12:59:34 +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
.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 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
Move min transmit bitrate to VideoEncoderConfig.
2014-10-24 09:23:21 +00:00
config.h
Move min transmit bitrate to VideoEncoderConfig.
2014-10-24 09:23:21 +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
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
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.