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nisse 085ec0e4b1 Delete sec_buffer.h. 
Unused since cl https://codereview.webrtc.org/1383253002

BUG=webrtc:6424

Review-Url: https://codereview.webrtc.org/2373483002
Cr-Commit-Position: refs/heads/master@{#14500}
2016-10-04 13:46:00 +00:00
..
androidjunit
Format all Java in WebRTC.
2016-09-29 11:12:51 +00:00
api
RTCCertificateStats[1] added.
2016-10-03 21:17:03 +00:00
audio
Restarting channel when swapping AudioReceiveStreams in WebrtcVoE.
2016-10-04 09:45:54 +00:00
base
Delete sec_buffer.h.
2016-10-04 13:46:00 +00:00
build
Roll chromium_revision f86fb54ec3..dd442d4812 (420104:421425)
2016-09-28 07:45:12 +00:00
call
Moved RtcEventLog files from call/ to logging/
2016-10-04 01:31:32 +00:00
common_audio
Revert of Test RTC_DCHECK_IS_ON instead of checking DCHECK_ALWAYS_ON everywhere (patchset #2 id:20001 of https://codereview.webrtc.org/2384693002/ )
2016-10-03 15:32:36 +00:00
common_video
Sort #includes that got unsorted when gmock.h and gtest.h moved to webrtc/test/
2016-10-01 05:29:53 +00:00
examples
Disable TCPChannelClientTest.testConnectIPv6
2016-09-30 07:20:12 +00:00
libjingle
GN: Change rtc_source_set targets --> rtc_static_library
2016-09-23 07:38:58 +00:00
logging
Moved RtcEventLog files from call/ to logging/
2016-10-04 01:31:32 +00:00
media
Restarting channel when swapping AudioReceiveStreams in WebrtcVoE.
2016-10-04 09:45:54 +00:00
modules
Centralize deactivation of Unequal Protection.
2016-10-04 12:28:05 +00:00
p2p
Emit SignalReadyToSend even for "presumed writable" connections.
2016-09-30 22:16:57 +00:00
pc
Delete unused class cricket::MediaSinkInterface, and mediasink.h.
2016-10-04 12:33:53 +00:00
sdk
Remove Crit::Scope lock by using atomic bool property.
2016-09-30 07:21:28 +00:00
stats
RTCCertificateStats[1] added.
2016-10-03 21:17:03 +00:00
system_wrappers
Sort #includes that got unsorted when gmock.h and gtest.h moved to webrtc/test/
2016-10-01 05:29:53 +00:00
test
Centralize deactivation of Unequal Protection.
2016-10-04 12:28:05 +00:00
tools
Moved RtcEventLog files from call/ to logging/
2016-10-04 01:31:32 +00:00
video
Fixed flaky VideoSendStreamTests after ViEEncoder changes
2016-10-04 08:47:05 +00:00
voice_engine
Moved RtcEventLog files from call/ to logging/
2016-10-04 01:31:32 +00:00
.gitignore
…
BUILD.gn
Delete bandwidthsmoother.h.
2016-10-04 12:58:35 +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
…
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
Enable the -Wundef warning for clang
2016-09-29 00:42:08 +00:00
common.gyp
Moving/renaming webrtc/common.h.
2016-09-07 14:34:45 +00:00
config.cc
Releand of Let ViEEncoder handle resolution changes.
2016-10-03 06:45:33 +00:00
config.h
Releand of Let ViEEncoder handle resolution changes.
2016-10-03 06:45:33 +00:00
DEPS
Enable the -Wundef warning for clang
2016-09-29 00:42:08 +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
…
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
…
PRESUBMIT.py
…
README.chromium
…
supplement.gypi
Cleanup MIPS specific link configuration
2016-06-20 05:39:25 +00:00
transport.h
…
typedefs.h
[MIPS] Fix build issue for mips64el
2016-06-23 10:52:35 +00:00
video_decoder.h
…
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_frame.h
Releand of Let ViEEncoder handle resolution changes.
2016-10-03 06:45:33 +00:00
video_receive_stream.h
Add logging statements to places where the frame might be dropped in WebRTC pipeline.
2016-09-30 13:19:12 +00:00
video_send_stream.h
Add VideoSendStream::Stats::prefered_media_bitrate_bps
2016-09-29 09:49:01 +00:00
webrtc_examples.gyp
Reland of name AppRTCDemo on Android and iOS to AppRTCMobile (patchset #1 id:1 of https://codereview.webrtc.org/2358133003/ )
2016-09-27 08:10:52 +00:00
webrtc_tests.gypi
GYP: Remove targets inside include_tests==1 that are converted to GN.
2016-09-15 11:57:39 +00:00
webrtc.gyp
Moved RtcEventLog files from call/ to logging/
2016-10-04 01:31:32 +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.