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braveyao@webrtc.org 9c89e932c9 WebRTCDemo: set local SSRC for loopback test, otherwise receiver would reset it due to ssrc clash, which would cause delayed remote rendering. 
(Including another fix here, https://review.webrtc.org/16779004/, to make the test run)

BUG=3500
TEST=Manual Test
R=glaznev@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/13819004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@6587 4adac7df-926f-26a2-2b94-8c16560cd09d
2014-07-03 05:59:22 +00:00
..
base webrtc/base: add dependent setting for gtest include directory that was missed when creating base_tests.gyp. Same as https://code.google.com/p/webrtc/source/browse/trunk/talk/libjingle_tests.gyp?r=6484#39 2014-07-01 16:39:17 +00:00
build Make deadlock suppressions less generic. 2014-07-02 14:19:05 +00:00
common_audio GN: Add BUILD.gn files + kjellander to OWNERS 2014-06-23 19:21:07 +00:00
common_video GN: Add BUILD.gn files + kjellander to OWNERS 2014-06-23 19:21:07 +00:00
examples WebRTCDemo: set local SSRC for loopback test, otherwise receiver would reset it due to ssrc clash, which would cause delayed remote rendering. 2014-07-03 05:59:22 +00:00
modules Implement BUILD.gn for desktop_capture. 2014-07-02 15:47:12 +00:00
overrides/webrtc/base Rename webrtc/base's IS_ALIGNED macro to RTC_IS_ALIGNED to avoid conflict between webrtc/base/basictypes.h and third_party/.../vpx_codec.h. 2014-05-21 16:52:14 +00:00
system_wrappers GN: Fix build by disabling compiler warning in base. 2014-06-29 13:37:08 +00:00
test Reserve RTP/RTCP modules in SetSSRC. 2014-06-30 13:19:09 +00:00
tools Pass GYP DEPTH variable to isolate. 2014-06-13 09:02:15 +00:00
video Reserve RTP/RTCP modules in SetSSRC. 2014-06-30 13:19:09 +00:00
video_engine Possibly fix deadlock happening due to unregister/register modules as switching between AST and TSO estimators. 2014-07-02 13:23:19 +00:00
voice_engine Add ExperimentalNs support in Config 2014-06-30 17:39:53 +00:00
.gitignore .gitignore: Add *.mk, created as part of ChromiumOS build 2013-01-04 21:25:42 +00:00
BUILD.gn GN: Refactor base/BUILD.gn and fix dbus-glib error. 2014-06-28 18:05:22 +00:00
call.h Implements start bitrate for new video API. 2014-06-16 08:57:39 +00:00
common_types.h Add ToString() to VideoSendStream::Config. 2014-05-15 09:35:06 +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 Add ToString() to VideoSendStream::Config. 2014-05-15 09:35:06 +00:00
engine_configurations.h Removes parts of the webrtc::VoEExternalMedia sub API as part of a clean-up operation where the goal is to remove unused APIs. 2014-05-12 12:19:19 +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
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
supplement.gypi TSan: Move suppressions to source file. 2014-06-27 09:18:51 +00:00
transport.h Rename newapi::Transport::SendRTP()->SendRtp(). 2013-11-20 12:17:04 +00:00
typedefs.h Remove ALLOW_UNUSED. 2014-05-05 18:18:02 +00:00
video_engine_tests.isolate Pass GYP DEPTH variable to isolate. 2014-06-13 09:02:15 +00:00
video_receive_stream.h Rename Start/Stop in Video{Send,Receive}Streams. 2014-04-24 11:13:21 +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 Enable pacing by default and remove the option to disable it from the new API. 2014-06-12 15:12:25 +00:00
webrtc_examples.gyp WebRTCDemo: set local SSRC for loopback test, otherwise receiver would reset it due to ssrc clash, which would cause delayed remote rendering. 2014-07-03 05:59:22 +00:00
webrtc_perf_tests.isolate Pass GYP DEPTH variable to isolate. 2014-06-13 09:02:15 +00:00
webrtc_tests.gypi Refactor Call-based tests. 2014-06-27 08:47:52 +00:00
webrtc.gyp Add ToString() to VideoSendStream::Config. 2014-05-15 09:35:06 +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.