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platform-external-webrtc/webrtc
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Peter Thatcher 3e95d3ef39 Don't log warning for unexpected STUN binding responses. 
It was too spammy in the log because we have many code paths that check for responses when it's not a problem that it's not an expected response.

R=glaznev@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/47199004

Cr-Commit-Position: refs/heads/master@{#9212}
2015-05-18 22:55:06 +00:00
..
base
Move SetCurrentThreadName to platform_thread.* in rtc_base_approved,
2015-05-18 11:50:31 +00:00
build
Adding support for OpenSL ES output in native WebRTC
2015-05-18 14:49:04 +00:00
common_audio
Add 3-band filter-bank implementation
2015-04-27 18:34:16 +00:00
common_video
Move IncomingVideoFrames to common_video/.
2015-05-13 11:28:04 +00:00
examples/android/media_demo
Adding support for OpenSL ES output in native WebRTC
2015-05-18 14:49:04 +00:00
libjingle
GYP variables for building expat, icu, libsrtp, usrsctp
2015-04-27 12:38:37 +00:00
modules
Make the BlockDifference() functions return DiffInfo as their callers expect.
2015-05-18 21:39:07 +00:00
overrides
Further updates to fix libjingle logging.
2015-05-14 02:18:15 +00:00
p2p
Don't log warning for unexpected STUN binding responses.
2015-05-18 22:55:06 +00:00
sound
Remove henrike@ from OWNERS
2015-04-01 15:08:49 +00:00
system_wrappers
Move SetCurrentThreadName to platform_thread.* in rtc_base_approved,
2015-05-18 11:50:31 +00:00
test
Remove VideoEngine interfaces.
2015-05-12 14:51:08 +00:00
tools
Disables mic bump-up level if not built with chromium
2015-04-22 04:39:47 +00:00
video
Add HW fallback option to software decoding.
2015-05-18 17:41:50 +00:00
video_engine
Remove some dead code in ViEChannel.
2015-05-18 09:48:17 +00:00
voice_engine
Adding support for OpenSL ES output in native WebRTC
2015-05-18 14:49:04 +00:00
.gitignore
…
audio_receive_stream.h
Add AudioReceiveStream to Call API.
2015-04-29 13:24:10 +00:00
BUILD.gn
Adding support for OpenSL ES output in native WebRTC
2015-05-18 14:49:04 +00:00
call.h
Add AudioReceiveStream to Call API.
2015-04-29 13:24:10 +00:00
codereview.settings
Add codereview.settings to the /webrtc subdirectory
2014-12-05 13:43:35 +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
…
config.cc
Add AudioReceiveStream to Call API.
2015-04-29 13:24:10 +00:00
config.h
Reland r9159 "Adding a new constraint to set NetEq buffer capacity ..."
2015-05-11 10:44:20 +00:00
engine_configurations.h
Remove VideoEngine interfaces.
2015-05-12 14:51:08 +00:00
experiments.h
Remove no longer used SkipEncodingUnusedStreams.
2014-07-22 07:17:17 +00:00
frame_callback.h
Changing include guard in frame_callback.h.
2015-02-03 14:51:39 +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
Update all .isolate files for the new format.
2014-10-31 18:08:09 +00:00
supplement.gypi
Move target_subarch from gyp_webrtc to supplement.gypi
2015-04-16 07:24:23 +00:00
transport.h
…
typedefs.h
Roll chromium_revision e144d30..6fdb142 (318658:318841) + remove OVERRIDE macro
2015-03-04 13:04:54 +00:00
video_decoder.h
Add HW fallback option to software decoding.
2015-05-18 17:41:50 +00:00
video_encoder.h
Dynamic resolution change for VP8 HW encode.
2015-04-21 22:29:53 +00:00
video_engine_tests.isolate
Update isolate files for Android APK tests.
2014-11-13 08:35:05 +00:00
video_frame.h
Reland "Remove usage of webrtc::NativeHandle since is just adds an extra level of indirection.""
2015-04-09 11:44:19 +00:00
video_receive_stream.h
Use IncomingVideoStream in VideoReceiveStream.
2015-04-24 13:15:40 +00:00
video_renderer.h
Use VideoReceiveStream as an ExternalRenderer.
2015-02-09 15:15:24 +00:00
video_send_stream.h
Reland "Make the entry point for VideoFrames to webrtc const ref I420VideoFrame.""
2015-03-18 09:51:44 +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
Add HW fallback option to software decoding.
2015-05-18 17:41:50 +00:00
webrtc.gyp
Rename GYP and GN targets for video capture+render.
2015-02-11 07:47:47 +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.