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platform-external-webrtc/webrtc
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Zeke Chin d830aeafe9 Add tkchin to video_coding OWNERS. 
BUG=
R=stefan@webrtc.org

Review URL: https://codereview.webrtc.org/1217853006.

Cr-Commit-Position: refs/heads/master@{#9528}
2015-07-01 23:01:56 +00:00
..
base
Add AsyncInvoker files for chromium GN build
2015-06-22 06:37:09 +00:00
build
iOS HW H264 support.
2015-06-29 21:35:08 +00:00
common_audio
Remove a cast again, after it was shown to worsen Windows perf.
2015-06-29 22:16:48 +00:00
common_video
Re-land "Convert native handles to buffers before encoding."
2015-06-05 09:08:12 +00:00
examples/android/media_demo
VoE: Remove unused interfaces
2015-05-26 08:25:00 +00:00
libjingle
Remove webrtc/libjingle/{examples,session}.
2015-06-12 11:13:46 +00:00
modules
Add tkchin to video_coding OWNERS.
2015-07-01 23:01:56 +00:00
overrides
Attempt at fixing error on the Chrome Windows FYI bots.
2015-05-25 19:22:23 +00:00
p2p
Remove iostream which causes a new static initializer
2015-06-23 04:54:22 +00:00
sound
Remove henrike@ from OWNERS
2015-04-01 15:08:49 +00:00
system_wrappers
Restore rows() and cols() in aligned_array.h
2015-06-20 18:11:02 +00:00
test
Move frame input (ViECapturer) to webrtc/video/.
2015-06-26 04:58:23 +00:00
tools
Pull the Voice Activity Detector out from the AGC
2015-06-25 19:28:55 +00:00
video
Fail RTP parsing on excessive padding length.
2015-07-01 12:35:56 +00:00
video_engine
Move frame input (ViECapturer) to webrtc/video/.
2015-06-26 04:58:23 +00:00
voice_engine
Reformat existing code. There should be no functional effects.
2015-06-11 21:31:48 +00:00
.gitignore
…
audio_receive_stream.h
VoE2 API draft
2015-06-08 11:05:07 +00:00
audio_send_stream.h
VoE2 API draft
2015-06-08 11:05:07 +00:00
BUILD.gn
iOS HW H264 support.
2015-06-29 21:35:08 +00:00
call.h
C++11 in-class member initialization in Call configs.
2015-06-11 10:38:46 +00:00
codereview.settings
Update deeper codereview.settings files to match the root.
2015-06-24 19:40:39 +00:00
common_types.cc
Parsing of transport wide sequence number rtp extension header.
2015-03-17 14:33:46 +00:00
common_types.h
Parsing of transport wide sequence number rtp extension header.
2015-03-17 14:33:46 +00:00
common.gyp
Fix style violations in common_types.h and config.h
2015-02-26 14:01:28 +00:00
common.h
Make Config::default_value leak instead of having an exit-time destructor.
2015-05-23 00:50:33 +00:00
config.cc
Add AudioReceiveStream to Call API.
2015-04-29 13:24:10 +00:00
config.h
Add options for NetEq fast accelerate mode through libjingle
2015-06-01 08:29:55 +00:00
engine_configurations.h
Remove VideoEngine interfaces.
2015-05-12 14:51:08 +00:00
experiments.h
…
frame_callback.h
Rename I420VideoFrame to VideoFrame.
2015-05-30 00:21:56 +00:00
LICENSE
…
LICENSE_THIRD_PARTY
Add the Ooura FFT to RealFourier.
2015-03-19 20:07:43 +00:00
OWNERS
…
PATENTS
…
PRESUBMIT.py
…
README.chromium
…
rtc_unittests.isolate
…
supplement.gypi
Move target_subarch from gyp_webrtc to supplement.gypi
2015-04-16 07:24:23 +00:00
transport.h
…
typedefs.h
Clear ARM NEON flag
2015-05-20 05:20:04 +00:00
video_decoder.h
iOS HW H264 support.
2015-06-29 21:35:08 +00:00
video_encoder.h
iOS HW H264 support.
2015-06-29 21:35:08 +00:00
video_engine_tests.isolate
…
video_frame.h
Re-land "Convert native handles to buffers before encoding."
2015-06-05 09:08:12 +00:00
video_receive_stream.h
C++11 in-class member initialization in Call configs.
2015-06-11 10:38:46 +00:00
video_renderer.h
Rename I420VideoFrame to VideoFrame.
2015-05-30 00:21:56 +00:00
video_send_stream.h
Move frame input (ViECapturer) to webrtc/video/.
2015-06-26 04:58:23 +00:00
webrtc_examples.gyp
Adding support for OpenSL ES output in native WebRTC
2015-05-18 14:49:04 +00:00
webrtc_perf_tests.isolate
Enable GoogleWifiTrace3Mbps simulations.
2015-05-04 12:26:26 +00:00
webrtc_tests.gypi
Move frame input (ViECapturer) to webrtc/video/.
2015-06-26 04:58:23 +00:00
webrtc.gyp
Merge video_engine_core into webrtc target.
2015-06-22 05:57:26 +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.