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kthelgason 917a4eeb60 Replace SequencedTaskChecker in RTPSenderVideo 
This is not the right place for a SequencedTaskChecker, as we can
not make any guarantees about the thread this method runs on.
We were hitting this check on Android and iOS whenever the encoder
would be reconfigured. Access to these ivars should be guarded
by a lock.

As a bonus, an unused method declaration was removed.

BUG=webrtc:6686

Review-Url: https://codereview.webrtc.org/2495483002
Cr-Commit-Position: refs/heads/master@{#15019}
2016-11-10 14:22:23 +00:00
..
androidjunit
Format all Java in WebRTC.
2016-09-29 11:12:51 +00:00
api
Don't copy loop variables when we don't have to
2016-11-10 13:15:41 +00:00
audio
Set actual transport overhead in rtp_rtcp
2016-11-08 10:50:16 +00:00
base
Ping the premier connection on each network with higher priority.
2016-11-09 04:04:14 +00:00
build
Use bitcode by default for ios framework build.
2016-11-10 09:13:49 +00:00
call
Remove audio from BitrateEstimatorTest.
2016-11-08 11:45:01 +00:00
common_audio
GN: Exclude suppressions of Chromium Clang warnings for Chromium builds.
2016-10-17 06:56:20 +00:00
common_video
Add helper functions for negotiating H264 profile level id
2016-11-10 09:57:42 +00:00
examples
Add screenshare support to AppRTCMobile.
2016-11-09 15:37:32 +00:00
libjingle
GN: Move xmllite_xmpp_unittests into webrtc/libjingle
2016-11-08 11:18:43 +00:00
logging
GN: New conventions, default target and refactorings
2016-10-28 12:44:07 +00:00
media
Revert of Stop caching supported codecs in WebRtcVideoEngine2 (patchset #1 id:1 of https://codereview.webrtc.org/2492473002/ )
2016-11-10 11:36:57 +00:00
modules
Replace SequencedTaskChecker in RTPSenderVideo
2016-11-10 14:22:23 +00:00
p2p
First step in providing a UdpTransportChannel.
2016-11-09 21:44:13 +00:00
pc
Revert of Stop caching supported codecs in WebRtcVideoEngine2 (patchset #1 id:1 of https://codereview.webrtc.org/2492473002/ )
2016-11-10 11:36:57 +00:00
sdk
Bump mac deployment target to 10.9
2016-11-09 15:44:34 +00:00
stats
RTCMediaStream[Track]Stats added.
2016-11-08 14:29:26 +00:00
system_wrappers
Add stats for frequency offset when converting RTP timestamp to NTP time.
2016-11-02 14:14:10 +00:00
test
Revert of Extract bitrate allocation of spatial/temporal layers out of codec impl. (patchset #17 id:320001 of https://codereview.webrtc.org/2434073003/ )
2016-11-09 14:14:56 +00:00
tools
GN: New conventions, default target and refactorings
2016-10-28 12:44:07 +00:00
video
Declare VideoCodec.codec_specific_info private
2016-11-10 11:26:45 +00:00
voice_engine
Set actual transport overhead in rtp_rtcp
2016-11-08 10:50:16 +00:00
.gitignore
…
BUILD.gn
First step in providing a UdpTransportChannel.
2016-11-09 21:44:13 +00:00
call.h
Set actual transport overhead in rtp_rtcp
2016-11-08 10:50:16 +00:00
codereview.settings
…
common_types.cc
Declare VideoCodec.codec_specific_info private
2016-11-10 11:26:45 +00:00
common_types.h
Declare VideoCodec.codec_specific_info private
2016-11-10 11:26:45 +00:00
common.gyp
Delete old video defines in engine config.
2016-10-07 05:07:36 +00:00
config.cc
Declare VideoCodec.codec_specific_info private
2016-11-10 11:26:45 +00:00
config.h
Clean up logging in AudioSendStream::SetupSendCodec().
2016-10-25 18:19:11 +00:00
DEPS
Delete old video defines in engine config.
2016-10-07 05:07:36 +00:00
LICENSE
…
LICENSE_THIRD_PARTY
…
OWNERS
…
PATENTS
…
PRESUBMIT.py
…
README.chromium
…
supplement.gypi
…
transport.h
…
typedefs.h
…
video_decoder.h
Refactoring: Hide VideoCodec.codecSpecific as "private"
2016-10-25 16:05:15 +00:00
video_encoder.h
Revert of Extract bitrate allocation of spatial/temporal layers out of codec impl. (patchset #17 id:320001 of https://codereview.webrtc.org/2434073003/ )
2016-11-09 14:14:56 +00:00
video_frame.h
Revert of Delete all use of cricket::VideoFrame and cricket::WebRtcVideoFrame. (patchset #7 id:120001 of https://codereview.webrtc.org/2383093002/ )
2016-11-02 10:40:05 +00:00
video_receive_stream.h
Implement framesDecoded stat in video receive ssrc stats.
2016-10-26 14:09:29 +00:00
video_send_stream.h
Let ViEEncoder express resolution requests as Sinkwants.
2016-11-01 18:45:54 +00:00
voice_engine_configurations.h
Delete old video defines in engine config.
2016-10-07 05:07:36 +00:00
webrtc_examples.gyp
Moving stun_prober target from webrtc/p2p to webrtc/examples
2016-11-01 10:09:19 +00:00
webrtc_tests.gypi
…
webrtc.gyp
Clean up logging in AudioSendStream::SetupSendCodec().
2016-10-25 18:19:11 +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.