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Peter Boström 23914fe756 Reject RTP one-byte extension ID 0. 
Only accept local identifiers in the range 1-14 inclusive.

BUG=1788, chromium:471328
R=asapersson@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/50549004

Cr-Commit-Position: refs/heads/master@{#8900}
2015-03-31 13:08:13 +00:00
..
base
Add a lock to NSSContext to fix data race
2015-03-26 21:38:53 +00:00
build
Remove old suppression for ProcessThreadImpl.
2015-03-31 09:48:14 +00:00
common_audio
common_audio: Explicit cast in WebRtcSpl_NormW16 on ARM
2015-03-30 21:38:36 +00:00
common_video
Remove I420VideoFrame::SwapFrame
2015-03-24 12:43:40 +00:00
examples
Adding playout volume control to WebRtcAudioTrack.java.
2015-03-27 09:56:35 +00:00
libjingle
Use base/scoped_ptr.h; system_wrappers/interface/scoped_ptr.h is going away
2015-02-26 14:43:50 +00:00
modules
Avoid critsect for protection- and qm setting callbacks in VideoSender.
2015-03-31 13:07:26 +00:00
overrides
Split C++ class from macro overrides to fix Chromium build
2015-03-19 07:26:21 +00:00
p2p
Address comments from cr 43769004.
2015-03-24 16:56:34 +00:00
sound
Use std::min and std::max instead of self-defined functions such as rtc::_min/_max.
2015-02-12 11:55:32 +00:00
system_wrappers
Allow setting thread priorities in Chromium on all but linux platforms.
2015-03-23 20:11:45 +00:00
test
Split EventWrapper in twain.
2015-03-23 16:25:46 +00:00
tools
Roll chromium_revision e144d30..6fdb142 (318658:318841) + remove OVERRIDE macro
2015-03-04 13:04:54 +00:00
video
Reject RTP one-byte extension ID 0.
2015-03-31 13:08:13 +00:00
video_engine
Avoid critsect for protection- and qm setting callbacks in VideoSender.
2015-03-31 13:07:26 +00:00
voice_engine
Introduce CodecManager and move code from AudioCodingModuleImpl
2015-03-30 17:00:54 +00:00
.gitignore
…
BUILD.gn
Let Chromium declare the mips_dsp_rev build variable.
2015-03-04 09:51:17 +00:00
call.h
Moving the pacer and the pacer thread to ChannelGroup.
2015-03-26 10:11:22 +00:00
codereview.settings
Add codereview.settings to the /webrtc subdirectory
2014-12-05 13:43:35 +00:00
common_types.cc
Parsing of transport wide sequence number rtp extension header.
2015-03-17 14:33:46 +00:00
common_types.h
Parsing of transport wide sequence number rtp extension header.
2015-03-17 14:33:46 +00:00
common.gyp
Fix style violations in common_types.h and config.h
2015-02-26 14:01:28 +00:00
common.h
…
config.cc
Fix style violations in common_types.h and config.h
2015-02-26 14:01:28 +00:00
config.h
Add CVO support to Vie layer.
2015-03-12 20:51:50 +00:00
engine_configurations.h
Delete all codec-specific subclasses of ACMGenericCodec
2015-02-23 09:26:51 +00:00
experiments.h
…
frame_callback.h
Changing include guard in frame_callback.h.
2015-02-03 14:51:39 +00:00
LICENSE
…
LICENSE_THIRD_PARTY
Add the Ooura FFT to RealFourier.
2015-03-19 20:07:43 +00:00
OWNERS
…
PATENTS
…
PRESUBMIT.py
…
README.chromium
…
rtc_unittests.isolate
Update all .isolate files for the new format.
2014-10-31 18:08:09 +00:00
supplement.gypi
Roll chromium_revision 601e6f3..b0c3ed3 (315263:316737)
2015-02-18 10:38:11 +00:00
transport.h
…
typedefs.h
Roll chromium_revision e144d30..6fdb142 (318658:318841) + remove OVERRIDE macro
2015-03-04 13:04:54 +00:00
video_decoder.h
Revert "Changed argument occurences of const I420VideoFrame* to const I420VideoFrame& and non-const I420VideoFrame& to I420VideoFrame*."
2015-03-16 13:48:18 +00:00
video_encoder.h
Remove default arguments in EncodedImageCallback.
2015-02-09 09:14:48 +00:00
video_engine_tests.isolate
Update isolate files for Android APK tests.
2014-11-13 08:35:05 +00:00
video_frame.h
Remove I420VideoFrame::SwapFrame
2015-03-24 12:43:40 +00:00
video_receive_stream.h
Add decoder-timing stats to VideoReceiveStream.
2015-02-25 10:42:45 +00:00
video_renderer.h
Use VideoReceiveStream as an ExternalRenderer.
2015-02-09 15:15:24 +00:00
video_send_stream.h
Reland "Make the entry point for VideoFrames to webrtc const ref I420VideoFrame.""
2015-03-18 09:51:44 +00:00
webrtc_examples.gyp
Remove video from WebRTC Android example.
2015-03-25 13:32:30 +00:00
webrtc_perf_tests.isolate
Offline screenshare quality test, plus loopback.
2015-02-18 12:46:44 +00:00
webrtc_tests.gypi
rtc_unittests on Android
2015-03-04 08:48:17 +00:00
webrtc.gyp
Rename GYP and GN targets for video capture+render.
2015-02-11 07:47:47 +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.