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henrik.lundin@webrtc.org 6e95d7afab Increment RTP timestamps for padding packets 
This CL makes the padding packets get their own RTP timestamps,
rather than having the same timestamp as the last sent video
packet. The purpose is to solve Issue 2611, where the overuse-
detector does not react to padding packets.

A test was implemented to verify that the padding packets do
get their own timestamps.

BUG=2611
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/3869004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5125 4adac7df-926f-26a2-2b94-8c16560cd09d
2013-11-15 08:59:19 +00:00
..
build
Remove ".." from include_dirs in build/common.
2013-11-11 10:20:27 +00:00
common_audio
Upgrade scoped_ptr to Chromium's latest version.
2013-10-22 12:50:00 +00:00
common_video
Revert "Remove extra copy in VideoCaptureImpl::IncomingFrameI420"
2013-10-31 23:41:04 +00:00
modules
Increment RTP timestamps for padding packets
2013-11-15 08:59:19 +00:00
system_wrappers
Remove ".." from include_dirs in build/common.
2013-11-11 10:20:27 +00:00
test
Remove update_resources.py as it's no longer used.
2013-11-08 09:08:36 +00:00
tools
Remove update_resources.py as it's no longer used.
2013-11-08 09:08:36 +00:00
video
Implement VideoSendStream::SetCodec().
2013-11-14 08:58:14 +00:00
video_engine
Disable all vie_auto_tests on Linux for now (take 2)
2013-11-13 15:51:49 +00:00
voice_engine
Inject config when creating channels to override the existing one.
2013-11-13 00:02:48 +00:00
.gitignore
.gitignore: Add *.mk, created as part of ChromiumOS build
2013-01-04 21:25:42 +00:00
call_tests.cc
Separate Call API/build files from video_engine/.
2013-10-28 16:32:01 +00:00
call.cc
Implement VideoSendStream::SetCodec().
2013-11-14 08:58:14 +00:00
call.h
Implement VideoSendStream::SetCodec().
2013-11-14 08:58:14 +00:00
common_types.h
Move RtcpStatistics to webrtc/common_types.h, to be used by vie as well.
2013-10-28 09:21:07 +00:00
common_unittest.cc
Fix unnamed-type-template-args warnings on clang.
2013-06-27 10:18:09 +00:00
common.h
Add a Config class interface to AudioProcessing for passing options.
2013-07-25 18:28:29 +00:00
config.h
Separate Call API/build files from video_engine/.
2013-10-28 16:32:01 +00:00
engine_configurations.h
Adds a new voice engine warning for the typing noise off state.
2013-09-17 18:09:20 +00:00
frame_callback.h
Separate Call API/build files from video_engine/.
2013-10-28 16:32:01 +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
loopback.cc
Separate Call API/build files from video_engine/.
2013-10-28 16:32:01 +00:00
PATENTS
Move src/ -> webrtc/
2012-10-22 18:19:23 +00:00
PRESUBMIT.py
Modified the presubmit checks such that difference license templates are checked for in webrtc and talk folder.
2013-07-22 22:32:50 +00:00
README.chromium
Move src/ -> webrtc/
2012-10-22 18:19:23 +00:00
transport.h
Separate Call API/build files from video_engine/.
2013-10-28 16:32:01 +00:00
typedefs.h
Check if WARN_UNUSED_RESULT and COMPILE_ASSERT are defined.
2013-10-23 19:11:32 +00:00
video_engine_tests.isolate
Add missing dependencies to .isolate files
2013-11-05 14:28:57 +00:00
video_receive_stream.h
Separate Call API/build files from video_engine/.
2013-10-28 16:32:01 +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
Implement VideoSendStream::SetCodec().
2013-11-14 08:58:14 +00:00
webrtc_tests.gypi
Fix build/isolate.gypi path in webrtc_tests.gypi.
2013-10-28 18:10:29 +00:00
webrtc.gyp
Separate Call API/build files from video_engine/.
2013-10-28 16:32:01 +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.