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10022cdeae785187e1d4329042b4ed294c393a16
platform-external-webrtc/webrtc
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Magnus Jedvert 10022cdeae RTPFragmentationHeader::VerifyAndAllocateFragmentationHeader: Verify that size fits in 16 bits 
This CL change the function declaration from uint16_t to size_t, and CHECKs that the size fits in uint16_t before proceeding.

BUG=484432
R=tommi@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/47229004

Cr-Commit-Position: refs/heads/master@{#9246}
2015-05-21 09:40:35 +00:00
..
base
Make SHA1 computation thread-safe.
2015-05-20 18:25:21 +00:00
build
Remove unnecessary dependencies for voe when building with include_internal_audio_device==0.
2015-05-20 07:44:23 +00:00
common_audio
Clear ARM NEON flag
2015-05-20 05:20:04 +00:00
common_video
Move IncomingVideoFrames to common_video/.
2015-05-13 11:28:04 +00:00
examples/android/media_demo
Remove rtp_rtcp/ dump functionality.
2015-05-20 12:10:56 +00:00
libjingle
GYP variables for building expat, icu, libsrtp, usrsctp
2015-04-27 12:38:37 +00:00
modules
RTPFragmentationHeader::VerifyAndAllocateFragmentationHeader: Verify that size fits in 16 bits
2015-05-21 09:40:35 +00:00
overrides
Further updates to fix libjingle logging.
2015-05-14 02:18:15 +00:00
p2p
Allow setting maximum protocol version for SSL stream adapters.
2015-05-20 10:48:24 +00:00
sound
Remove henrike@ from OWNERS
2015-04-01 15:08:49 +00:00
system_wrappers
Move SetCurrentThreadName to platform_thread.* in rtc_base_approved,
2015-05-18 11:50:31 +00:00
test
Remove VideoEngine interfaces.
2015-05-12 14:51:08 +00:00
tools
Disables mic bump-up level if not built with chromium
2015-04-22 04:39:47 +00:00
video
Remove default encoder/decoders.
2015-05-21 07:56:17 +00:00
video_engine
Remove default encoder/decoders.
2015-05-21 07:56:17 +00:00
voice_engine
Remove rtp_rtcp/ dump functionality.
2015-05-20 12:10:56 +00:00
.gitignore
…
audio_receive_stream.h
Add AudioReceiveStream to Call API.
2015-04-29 13:24:10 +00:00
BUILD.gn
Clear ARM NEON flag
2015-05-20 05:20:04 +00:00
call.h
Add AudioReceiveStream to Call API.
2015-04-29 13:24:10 +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
Add AudioReceiveStream to Call API.
2015-04-29 13:24:10 +00:00
config.h
Reland r9159 "Adding a new constraint to set NetEq buffer capacity ..."
2015-05-11 10:44:20 +00:00
engine_configurations.h
Remove VideoEngine interfaces.
2015-05-12 14:51:08 +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
Move target_subarch from gyp_webrtc to supplement.gypi
2015-04-16 07:24:23 +00:00
transport.h
…
typedefs.h
Clear ARM NEON flag
2015-05-20 05:20:04 +00:00
video_decoder.h
Remove unused VideoDecoder methods.
2015-05-21 07:42:14 +00:00
video_encoder.h
Add HW fallback option to software encoding.
2015-05-19 21:09:17 +00:00
video_engine_tests.isolate
Update isolate files for Android APK tests.
2014-11-13 08:35:05 +00:00
video_frame.h
Reland "Remove usage of webrtc::NativeHandle since is just adds an extra level of indirection.""
2015-04-09 11:44:19 +00:00
video_receive_stream.h
Use IncomingVideoStream in VideoReceiveStream.
2015-04-24 13:15:40 +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
Adding support for OpenSL ES output in native WebRTC
2015-05-18 14:49:04 +00:00
webrtc_perf_tests.isolate
Enable GoogleWifiTrace3Mbps simulations.
2015-05-04 12:26:26 +00:00
webrtc_tests.gypi
Add HW fallback option to software encoding.
2015-05-19 21:09:17 +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.