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webrtc_m130/webrtc
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kthelgason dea7f4f46f Ignore aud and sei NALus when parsing bitstream. 
We currently don't know how to parse these NALus and we don't need
any information from them anyway so we might as well skip parsing them
and not break.

BUG=chromium:697795

Review-Url: https://codereview.webrtc.org/2732623002
Cr-Commit-Position: refs/heads/master@{#17057}
2017-03-06 10:49:36 +00:00
..
api
Reland "Enable GN check for webrtc/examples"
2017-03-06 08:29:21 +00:00
audio
Remove usage of VoEVolumeControl from WVoE and Audio[Send|Receive]Stream.
2017-03-02 01:02:23 +00:00
base
Fixing race between CallbackCanceled and CancelCallback in AsyncInvoker.
2017-03-03 18:33:18 +00:00
call
Add support for Location (RTC_FROM_HERE) to ProcessThread::RegisterModule.
2017-03-03 11:20:24 +00:00
common_audio
Enable GN check for webrtc/common_audio
2017-03-01 15:07:10 +00:00
common_video
Ignore aud and sei NALus when parsing bitstream.
2017-03-06 10:49:36 +00:00
examples
Add PESQ precompiled tool for audio quality testing
2017-03-06 10:23:34 +00:00
logging
Remove unused include from rtc_event_log_parser.cc
2017-02-28 09:46:19 +00:00
media
Add performance tracing for PlatformThread and parts of the video code.
2017-03-03 15:21:18 +00:00
modules
Fix memory leaks in Windows core audio
2017-03-06 10:24:42 +00:00
ortc
Revert of Enable GN check for webrtc/{ortc,p2p} (patchset #4 id:60001 of https://codereview.webrtc.org/2714263004/ )
2017-03-04 23:08:44 +00:00
p2p
Revert of Enable GN check for webrtc/{ortc,p2p} (patchset #4 id:60001 of https://codereview.webrtc.org/2714263004/ )
2017-03-04 23:08:44 +00:00
pc
Reland: Improve testing of SRTP external auth code paths.
2017-03-06 08:14:10 +00:00
sdk
Add unit tests for RTCMTLVideoView.
2017-03-03 16:24:06 +00:00
stats
Move RTCOutboundRTPStreamStats.roundTripTime to inbound, don't collect.
2017-03-01 09:02:45 +00:00
system_wrappers
Provide an inline implementation of IsEnabled.
2017-02-28 17:26:22 +00:00
test
Packet Loss Tracker - Stream Separation
2017-03-03 18:51:35 +00:00
tools
Add probe logging to RtcEventLog.
2017-02-27 10:18:46 +00:00
video
Add performance tracing for PlatformThread and parts of the video code.
2017-03-03 15:21:18 +00:00
voice_engine
Modify TransportFeedbackPacketLossTrackerTest to use parameterization
2017-03-03 19:11:06 +00:00
.gitignore
…
BUILD.gn
Revert of Removing unused audio device tests (patchset #1 id:1 of https://codereview.webrtc.org/2716843002/ )
2017-02-26 15:12:50 +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
Remove the Windows Wave audio device implementation.
2017-02-17 22:48:07 +00:00
config.cc
Adding "adapter" ORTC objects on top of ChannelManager/BaseChannel/etc.
2017-02-26 02:15:09 +00:00
config.h
Adding "adapter" ORTC objects on top of ChannelManager/BaseChannel/etc.
2017-02-26 02:15:09 +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
Reland of Add QP sum stats for received streams. (patchset #2 id:300001 of https://codereview.webrtc.org/2680893002/ )
2017-02-09 12:53:45 +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
Reland of Add QP sum stats for received streams. (patchset #2 id:300001 of https://codereview.webrtc.org/2680893002/ )
2017-02-09 12:53:45 +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
Reland of Setting is_component_build to false by default (patchset #1 id:1 of https://codereview.webrtc.org/2731703004/ )
2017-03-04 03:41:59 +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.