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webrtc_m130/webrtc
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deadbeef dd7fb43f28 Emit SignalReadyToSend even for "presumed writable" connections. 
The Connection class will now blindly forward SignalReadyToSend, and
P2PTransportChannel will decide whether to forward it further (which
it was already doing).

BUG=webrtc:6448

Review-Url: https://codereview.webrtc.org/2374183005
Cr-Commit-Position: refs/heads/master@{#14462}
2016-09-30 22:16:57 +00:00
..
androidjunit
Format all Java in WebRTC.
2016-09-29 11:12:51 +00:00
api
Add logging statements to places where the frame might be dropped in WebRTC pipeline.
2016-09-30 13:19:12 +00:00
audio
Enable the -Wundef warning for clang
2016-09-29 00:42:08 +00:00
base
Relanding: Allow the DTLS fingerprint verification to occur after the handshake.
2016-09-30 18:55:49 +00:00
build
Roll chromium_revision f86fb54ec3..dd442d4812 (420104:421425)
2016-09-28 07:45:12 +00:00
call
Add logging statements to places where the frame might be dropped in WebRTC pipeline.
2016-09-30 13:19:12 +00:00
common_audio
Enable the -Wundef warning for clang
2016-09-29 00:42:08 +00:00
common_video
Add logging statements to places where the frame might be dropped in WebRTC pipeline.
2016-09-30 13:19:12 +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
media
Reland of Delete VideoFrameFactory, CapturedFrame, and related code. (patchset #1 id:1 of https://codereview.webrtc.org/2357113002/ )
2016-09-30 08:34:35 +00:00
modules
Revert of Use sps and pps to determine decodability of H.264 frames. (patchset #4 id:60001 of https://codereview.webrtc.org/2341713002/ )
2016-09-30 16:06:43 +00:00
p2p
Emit SignalReadyToSend even for "presumed writable" connections.
2016-09-30 22:16:57 +00:00
pc
Update WebRTC to build against libsrtp 2.0
2016-09-28 21:08:53 +00:00
sdk
Remove Crit::Scope lock by using atomic bool property.
2016-09-30 07:21:28 +00:00
stats
GN: Change rtc_source_set targets --> rtc_static_library
2016-09-23 07:38:58 +00:00
system_wrappers
Add sanity check for decreasing RTP timestamp in RtpToNtpMs.
2016-09-30 10:16:26 +00:00
test
Add logging statements to places where the frame might be dropped in WebRTC pipeline.
2016-09-30 13:19:12 +00:00
tools
Reland: Fix race / crash in OnNetworkRouteChanged().
2016-09-30 08:07:00 +00:00
video
Add logging statements to places where the frame might be dropped in WebRTC pipeline.
2016-09-30 13:19:12 +00:00
voice_engine
Enable the -Wundef warning for clang
2016-09-29 00:42:08 +00:00
.gitignore
…
BUILD.gn
Disable TCPChannelClientTest.testConnectIPv6
2016-09-30 07:20:12 +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
Revert of Let ViEEncoder handle resolution changes. (patchset #17 id:340001 of https://codereview.webrtc.org/2351633002/ )
2016-09-30 06:25:46 +00:00
config.h
Revert of Let ViEEncoder handle resolution changes. (patchset #17 id:340001 of https://codereview.webrtc.org/2351633002/ )
2016-09-30 06:25:46 +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
Revert of Let ViEEncoder handle resolution changes. (patchset #17 id:340001 of https://codereview.webrtc.org/2351633002/ )
2016-09-30 06:25:46 +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
Move webrtc/audio_*.h to webrtc/api/call
2016-08-31 14:33:14 +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.