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webrtc_m130/webrtc
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sakal bb58435da0 Fix potential synchronization issues with framelisteners in EglRenderer. 
Previously, a frame queued before calling addFrameListener could be
passed to the listener. Also fixes an issue where listener could still
be called after removeFrameListener call returned.

BUG=webrtc:6470

Review-Url: https://codereview.webrtc.org/2529313002
Cr-Commit-Position: refs/heads/master@{#15275}
2016-11-28 16:53:50 +00:00
..
androidjunit
Format all Java in WebRTC.
2016-09-29 11:12:51 +00:00
api
Fix potential synchronization issues with framelisteners in EglRenderer.
2016-11-28 16:53:50 +00:00
audio
Revert of Pass time constant to bwe smoothing filter. (patchset #8 id:140001 of https://codereview.webrtc.org/2518923003/ )
2016-11-28 16:05:23 +00:00
base
Roll chromium_revision 5e821a778b..5c22c2afac (432715:434448)
2016-11-28 10:04:45 +00:00
build
Make Valgrind memcheck work in swarming.
2016-11-25 13:49:00 +00:00
call
Revert of Pass time constant to bwe smoothing filter. (patchset #8 id:140001 of https://codereview.webrtc.org/2518923003/ )
2016-11-28 16:05:23 +00:00
common_audio
Smooth BWE and pass it to Audio Network Adaptor.
2016-11-28 10:34:23 +00:00
common_video
Add I420Buffer::Copy method taking plane pointers as input.
2016-11-28 11:48:01 +00:00
examples
Reland of Allow custom metrics implementations on Android. (patchset #1 id:1 of https://codereview.webrtc.org/2516403002/ )
2016-11-23 15:12:28 +00:00
libjingle
Remove all references to GYP
2016-11-16 19:11:38 +00:00
logging
GN: New conventions, default target and refactorings
2016-10-28 12:44:07 +00:00
media
Pass selected cricket::VideoCodec down to internal H264 encoder
2016-11-28 15:20:26 +00:00
modules
Now run EndToEndTest with the WebRTC-NewVideoJitterBuffer experiment.
2016-11-28 16:49:15 +00:00
p2p
Remove all references to GYP
2016-11-16 19:11:38 +00:00
pc
Don't cache video codec list in VideoEngine2.
2016-11-28 14:02:28 +00:00
sdk
iOS HW encoder: Enable H264 High profile support
2016-11-25 15:32:37 +00:00
stats
Change rtc::TimeNanos and rtc::TimeMicros return value from uint64_t to int64_t.
2016-11-28 09:55:05 +00:00
system_wrappers
Use NtpTime in RtcpMeasurement instead of uint sec/uint frac.
2016-11-17 10:27:20 +00:00
test
Revert of Pass time constant to bwe smoothing filter. (patchset #8 id:140001 of https://codereview.webrtc.org/2518923003/ )
2016-11-28 16:05:23 +00:00
tools
Revert of Pass time constant to bwe smoothing filter. (patchset #8 id:140001 of https://codereview.webrtc.org/2518923003/ )
2016-11-28 16:05:23 +00:00
video
Now run EndToEndTest with the WebRTC-NewVideoJitterBuffer experiment.
2016-11-28 16:49:15 +00:00
voice_engine
Revert of Pass time constant to bwe smoothing filter. (patchset #8 id:140001 of https://codereview.webrtc.org/2518923003/ )
2016-11-28 16:05:23 +00:00
.gitignore
…
BUILD.gn
Moved transport.h from webrtc/ to webrtc/api, created build target and updated WebRTC dependencies.
2016-11-28 15:02:19 +00:00
call.h
Set actual transport overhead in rtp_rtcp
2016-11-08 10:50:16 +00:00
codereview.settings
…
common_types.cc
Unify VideoCodecType to/from string functionality
2016-11-22 18:17:04 +00:00
common_types.h
Reland of Add H264 profile to webrtc::VideoCodecH264 and webrtc::VideoPayload (patchset #1 id:1 of https://codereview.webrtc.org/2529143002/ )
2016-11-25 18:06:35 +00:00
config.cc
Keep all codec parameters in VideoReceiveStream::Decoder
2016-11-25 11:56:41 +00:00
config.h
Keep all codec parameters in VideoReceiveStream::Decoder
2016-11-25 11:56:41 +00:00
DEPS
Delete old video defines in engine config.
2016-10-07 05:07:36 +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
Remove all references to GYP
2016-11-16 19:11:38 +00:00
PATENTS
…
PRESUBMIT.py
…
README.chromium
…
transport.h
Moved transport.h from webrtc/ to webrtc/api, created build target and updated WebRTC dependencies.
2016-11-28 15:02:19 +00:00
typedefs.h
[MIPS] Fix build issue for mips64el
2016-06-23 10:52:35 +00:00
video_decoder.h
Move VideoDecoderSoftwareFallbackWrapper from webrtc/video_decoder.h to webrtc/media/engine/
2016-11-22 09:43:06 +00:00
video_encoder.h
Reland of Stop using hardcoded payload types for video codecs (patchset #1 id:1 of https://codereview.webrtc.org/2513633002/ )
2016-11-18 09:34:14 +00:00
video_frame.h
Reland of Delete all use of cricket::VideoFrame and cricket::WebRtcVideoFrame. (patchset #1 id:1 of https://codereview.webrtc.org/2471783002/ )
2016-11-11 11:55:19 +00:00
video_receive_stream.h
Moved transport.h from webrtc/ to webrtc/api, created build target and updated WebRTC dependencies.
2016-11-28 15:02:19 +00:00
video_send_stream.h
Moved transport.h from webrtc/ to webrtc/api, created build target and updated WebRTC dependencies.
2016-11-28 15:02:19 +00:00
voice_engine_configurations.h
Delete old video defines in engine config.
2016-10-07 05:07:36 +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.