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Danil Chapovalov a64a2fbf6d Fix oversized rtp extension parsing. 
When size of individual one-byte extension span beyound extension block

BUG=chromium:645201
R=brandtr@webrtc.org

Review URL: https://codereview.webrtc.org/2327743003 .

Cr-Commit-Position: refs/heads/master@{#14183}
2016-09-12 09:41:44 +00:00
..
androidjunit
Re-order and remove unused Java imports.
2016-08-02 13:44:19 +00:00
api
Add statistics for the time it takes to start and stop the camera on Camera2.
2016-09-12 08:05:28 +00:00
audio
OWNERS: Make everyone able to change *.gn,*.gni files.
2016-09-09 12:51:48 +00:00
base
Adding ability to simulate EWOULDBLOCK/SignalReadyToSend.
2016-09-09 20:16:25 +00:00
build
GN: Revert to default compiler optimizations for Win Release.
2016-09-12 06:29:49 +00:00
call
Remove RTC_LOGGED_* macro.
2016-09-10 05:40:34 +00:00
common_audio
OWNERS: Make everyone able to change *.gn,*.gni files.
2016-09-09 12:51:48 +00:00
common_video
Revert "Optimize Android NV12 capture"
2016-09-09 23:15:15 +00:00
examples
OWNERS: Make everyone able to change *.gn,*.gni files.
2016-09-09 12:51:48 +00:00
libjingle
OWNERS: Make everyone able to change *.gn,*.gni files.
2016-09-09 12:51:48 +00:00
media
Revert of Introduced new scheme for controlling the functionality inside the audio processing module (patchset #12 id:260001 of https://codereview.webrtc.org/2292863002/ )
2016-09-12 06:04:37 +00:00
modules
Fix oversized rtp extension parsing.
2016-09-12 09:41:44 +00:00
p2p
Removing assert error when we fail to create a connection for a ping from an unknown address.
2016-09-10 03:48:08 +00:00
pc
OWNERS: Make everyone able to change *.gn,*.gni files.
2016-09-09 12:51:48 +00:00
sdk
OWNERS: Make everyone able to change *.gn,*.gni files.
2016-09-09 12:51:48 +00:00
stats
OWNERS: Make everyone able to change *.gn,*.gni files.
2016-09-09 12:51:48 +00:00
system_wrappers
Remove RTC_LOGGED_* macro.
2016-09-10 05:40:34 +00:00
test
Change owner of webrtc/test/channel_transport to solenberg@.
2016-09-08 15:41:59 +00:00
tools
Don't use VoE legacy APIs in force_mic_volume_max tool.
2016-09-09 18:05:01 +00:00
video
Add periodic logging of number of captured and dropped frames in VideoCaptureInput. Logged every minute.
2016-09-12 07:10:53 +00:00
voice_engine
OWNERS: Make everyone able to change *.gn,*.gni files.
2016-09-09 12:51:48 +00:00
.gitignore
Flesh out webrtc/.gitignore
2015-11-16 19:02:02 +00:00
BUILD.gn
Reland of move all reference to carbon api (patchset #1 id:1 of https://codereview.webrtc.org/2317343003/ )
2016-09-09 10:19:54 +00:00
call.h
This reverts commit 8eb37a39e79fe1098d3503dcb8c8c2d196203fed. Chrome now have its own implementation of TaskQueues that is based on Chrome threads.
2016-09-01 08:17:43 +00:00
codereview.settings
Give a more specific URL for creating WebRTC checkout
2016-04-01 07:36:53 +00:00
common_types.cc
Reland of Ignore Camera and Flip bits in CVO when parsing video rotation (patchset #1 id:1 of https://codereview.webrtc.org/2300323002/ )
2016-09-08 10:25:05 +00:00
common_types.h
Reland of Ignore Camera and Flip bits in CVO when parsing video rotation (patchset #1 id:1 of https://codereview.webrtc.org/2300323002/ )
2016-09-08 10:25:05 +00:00
common.gyp
Moving/renaming webrtc/common.h.
2016-09-07 14:34:45 +00:00
config.cc
Avoid unnecessary HW video encoder reconfiguration
2016-06-16 19:08:11 +00:00
config.h
Moving/renaming webrtc/common.h.
2016-09-07 14:34:45 +00:00
DEPS
Reland of Ignore Camera and Flip bits in CVO when parsing video rotation (patchset #1 id:1 of https://codereview.webrtc.org/2300323002/ )
2016-09-08 10:25:05 +00:00
engine_configurations.h
Reland of move all reference to carbon api (patchset #1 id:1 of https://codereview.webrtc.org/2317343003/ )
2016-09-09 10:19:54 +00:00
LICENSE
Move src/ -> webrtc/
2012-10-22 18:19:23 +00:00
LICENSE_THIRD_PARTY
Delete talk directory, and references from build_ios_libs.sh.
2016-09-02 10:07:13 +00:00
OWNERS
OWNERS: Make everyone able to change *.gn,*.gni files.
2016-09-09 12:51:48 +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_apk.isolate
Add isolate files for Android tests
2016-04-18 03:08:28 +00:00
rtc_unittests.isolate
Update all .isolate files for the new format.
2014-10-31 18:08:09 +00:00
supplement.gypi
Cleanup MIPS specific link configuration
2016-06-20 05:39:25 +00:00
transport.h
Reland of Merge webrtc/video_engine/ into webrtc/video/ (patchset #2 id:300001 of https://codereview.webrtc.org/1507903005/ )
2015-12-09 11:13:40 +00:00
typedefs.h
[MIPS] Fix build issue for mips64el
2016-06-23 10:52:35 +00:00
video_decoder.h
Replace scoped_ptr with unique_ptr in webrtc/common_video/
2016-03-02 11:41:40 +00:00
video_encoder.h
Add codec name to CodecSpecificInfo and get the codec name stats from there instead.
2016-09-01 07:21:18 +00:00
video_engine_tests_apk.isolate
Add isolate files for Android tests
2016-04-18 03:08:28 +00:00
video_engine_tests.isolate
Add audio streams to CallTest and a first A/V call test.
2016-01-07 16:43:31 +00:00
video_frame.h
Introduce webrtc::VideoFrame::timestamp_us, and corresponding constructor.
2016-09-05 07:51:25 +00:00
video_receive_stream.h
Add periodic logging of video stats.
2016-08-11 15:41:26 +00:00
video_send_stream.h
This reverts commit 8eb37a39e79fe1098d3503dcb8c8c2d196203fed. Chrome now have its own implementation of TaskQueues that is based on Chrome threads.
2016-09-01 08:17:43 +00:00
webrtc_examples.gyp
Delete talk directory, and references from build_ios_libs.sh.
2016-09-02 10:07:13 +00:00
webrtc_nonparallel_tests_apk.isolate
Add isolate files for Android tests
2016-04-18 03:08:28 +00:00
webrtc_nonparallel_tests.isolate
Create isolate files for nonparallel tests.
2015-10-29 14:29:15 +00:00
webrtc_perf_tests_apk.isolate
Add isolate files for Android tests
2016-04-18 03:08:28 +00:00
webrtc_perf_tests.isolate
Add an HD resolution perf test.
2016-08-11 14:01:03 +00:00
webrtc_tests.gypi
Reland of move all reference to carbon api (patchset #1 id:1 of https://codereview.webrtc.org/2317343003/ )
2016-09-09 10:19:54 +00:00
webrtc.gyp
Move webrtc/audio_*.h to webrtc/api/call
2016-08-31 14:33:14 +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.