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ivoc a48e1b6bc3 Fix for left shift of potentially negative value. 
Left shifting of negative integers is undefined behavior, and should be prevented. This CL fixes one such instance in the Levinson Durbin function.

BUG=chromium:681377

Review-Url: https://codereview.webrtc.org/2680973005
Cr-Commit-Position: refs/heads/master@{#16507}
2017-02-09 11:05:59 +00:00
..
api
Adding more comments to every header file in api/ subdirectory.
2017-02-08 09:38:21 +00:00
audio
Wire up audio packet loss to BWE.
2017-02-07 15:14:08 +00:00
base
Use fake clock in some more networks tests.
2017-02-08 21:18:00 +00:00
call
Move RemoteBitrateEstimator::RemoveStream calls from receive streams to Call.
2017-02-07 09:18:43 +00:00
common_audio
Fix for left shift of potentially negative value.
2017-02-09 11:05:59 +00:00
common_video
Delete VideoReceiveStream::Config::pre_render_callback.
2017-02-07 17:37:41 +00:00
examples
Move Android video quality loopback script.
2017-02-09 09:05:54 +00:00
logging
Add option to print information about configured SSRCs from RTC event logs.
2017-02-08 17:37:30 +00:00
media
Revert of Add QP sum stats for received streams. (patchset #10 id:180001 of https://codereview.webrtc.org/2649133005/ )
2017-02-07 18:59:25 +00:00
modules
Log information (at level LS_INFO) about which overuse estimator is used.
2017-02-08 16:45:20 +00:00
p2p
Increase STUN RTOs to work better on poor networks, such as 2G networks.
2017-02-08 22:42:22 +00:00
pc
Increase STUN RTOs to work better on poor networks, such as 2G networks.
2017-02-08 22:42:22 +00:00
sdk
Roll chromium_revision cf2dce6a6d..2019b9e075 (448969:449230)
2017-02-09 07:56:43 +00:00
stats
RTCRTPStreamStats.mediaTrackId renamed to trackId.
2017-01-27 14:35:16 +00:00
system_wrappers
Reland of Moving webrtc.gni up one level from build/ (patchset #1 id:1 of https://codereview.webrtc.org/2657563002/ )
2017-01-24 14:58:22 +00:00
test
Add new target direct_transport and remove fake_network and direct_transport from test_common.
2017-02-08 13:55:51 +00:00
tools
Move Android video quality loopback script.
2017-02-09 09:05:54 +00:00
video
Fix for flaky RemoveOverheadFromBandwidth test.
2017-02-08 16:21:52 +00:00
voice_engine
Remove unused voe_output_test.cc.
2017-02-09 09:23:25 +00:00
.gitignore
…
BUILD.gn
Added gn target for rtc_event_log2rtp_dump.
2017-02-08 08:28:09 +00:00
call.h
Moved call.h and most of api/call/* into call/
2016-12-07 12:53:04 +00:00
codereview.settings
…
common_types.cc
Reject XR TargetBitrate items with unsupported layer indices
2016-12-06 14:09:00 +00:00
common_types.h
Move VideoFrame and related declarations to webrtc/api/video.
2017-01-10 15:44:26 +00:00
config.cc
Remove FlexfecConfig and replace with specific struct in VideoSendStream.
2017-01-16 14:59:19 +00:00
config.h
Remove FlexfecConfig and replace with specific struct in VideoSendStream.
2017-01-16 14:59:19 +00:00
DEPS
Move VideoFrame and related declarations to webrtc/api/video.
2017-01-10 15:44:26 +00:00
LICENSE
…
LICENSE_THIRD_PARTY
Delete unused spreadsort implementation.
2016-12-05 11:03:26 +00:00
no_op_function.cc
Moving no_op_function.cc out of webrtc/build
2017-01-24 13:25:12 +00:00
OWNERS
Cleanup unused rules in webrtc/DEPS + add kjellander to OWNERS for it
2016-11-29 10:52:22 +00:00
PATENTS
…
PRESUBMIT.py
…
README.chromium
…
typedefs.h
Only define NO_RETURN if undefined
2017-01-30 08:54:19 +00:00
video_decoder.h
Revert of Add QP sum stats for received streams. (patchset #10 id:180001 of https://codereview.webrtc.org/2649133005/ )
2017-02-07 18:59:25 +00:00
video_encoder.h
Move usage of QualityScaler to ViEEncoder.
2016-11-29 09:44:22 +00:00
video_frame.h
Move VideoFrame and related declarations to webrtc/api/video.
2017-01-10 15:44:26 +00:00
video_receive_stream.h
Revert of Add QP sum stats for received streams. (patchset #10 id:180001 of https://codereview.webrtc.org/2649133005/ )
2017-02-07 18:59:25 +00:00
video_send_stream.h
Remove FlexfecConfig and replace with specific struct in VideoSendStream.
2017-01-16 14:59:19 +00:00
webrtc.gni
Adding build switch for Opus that supports 120ms ptime.
2017-02-02 01:31:11 +00:00
whitespace.txt
Moving whitespace file up by one folder
2017-01-24 13:19:42 +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.