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stefan 12a39f4100 Don't crash on unexpected stap-a or fu-a. 
BUG=chromium:655091

Review-Url: https://codereview.webrtc.org/2406363004
Cr-Commit-Position: refs/heads/master@{#14618}
2016-10-12 22:30:18 +00:00
..
androidjunit
Format all Java in WebRTC.
2016-09-29 11:12:51 +00:00
api
DataChannelInterface default impl of [messages/bytes]_[sent/received].
2016-10-12 21:48:20 +00:00
audio
Add RtcpRttStats to AudioStream
2016-10-11 07:29:34 +00:00
base
Remove deprecated Gestalt methods.
2016-10-10 18:19:22 +00:00
build
GN: Build iOS framework in build_ios_libs.sh
2016-10-11 20:27:34 +00:00
call
Add RtcpRttStats to AudioStream
2016-10-11 07:29:34 +00:00
common_audio
Fix "left shift of negative value" bug
2016-10-07 14:38:54 +00:00
common_video
Revert of Delete webrtc::VideoFrame::CopyFrame. (patchset #2 id:20001 of https://codereview.webrtc.org/2371363003/ )
2016-10-06 13:29:34 +00:00
examples
Speculatively increase timeouts for DirectRTCClientTest to reduce flakiness.
2016-10-06 07:34:47 +00:00
libjingle
GN: Change rtc_source_set targets --> rtc_static_library
2016-09-23 07:38:58 +00:00
logging
Added logging for audio send/receive stream configs.
2016-10-10 12:12:57 +00:00
media
Removed the legacy behavior of stopping playout when setting new receive codecs.
2016-10-12 10:20:34 +00:00
modules
Don't crash on unexpected stap-a or fu-a.
2016-10-12 22:30:18 +00:00
p2p
Revert of Prune connections based on network name. (patchset #3 id:130001 of https://codereview.webrtc.org/2395243005/ )
2016-10-11 13:43:36 +00:00
pc
Remove useless debugging code
2016-10-10 22:57:00 +00:00
sdk
Using relative path for GN for iOS.
2016-10-11 23:12:47 +00:00
stats
RTCIceCandidatePairStats[1] added.
2016-10-11 21:54:55 +00:00
system_wrappers
Fixed flaky clock_unittest by using relative comparison.
2016-10-05 08:55:10 +00:00
test
Add RtcpRttStats to AudioStream
2016-10-11 07:29:34 +00:00
tools
Fix delay plot crash in event_log_visualizer.
2016-10-10 15:16:33 +00:00
video
Set screenshare.enabled parameter to false when running video_loopback test.
2016-10-11 23:20:40 +00:00
voice_engine
Voe::Channel: Turned GetPlayoutFrequency into GetRtpTimestampRateHz.
2016-10-12 18:04:16 +00:00
.gitignore
…
BUILD.gn
Delete macsocketserver.h and related files.
2016-10-07 12:57:26 +00:00
call.h
Refactoring: move ownership of RtcEventLog from Call to PeerConnection
2016-10-07 18:53:15 +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
Delete old video defines in engine config.
2016-10-07 05:07:36 +00:00
config.cc
Rename FecConfig to UlpfecConfig in config.h.
2016-10-05 06:28:43 +00:00
config.h
Rename FecConfig to UlpfecConfig in config.h.
2016-10-05 06:28:43 +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
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
Make MediaCodecEncoder fallback to a software encoder on failure.
2016-10-06 12:55:19 +00:00
video_frame.h
Revert of Delete webrtc::VideoFrame::CopyFrame. (patchset #2 id:20001 of https://codereview.webrtc.org/2371363003/ )
2016-10-06 13:29:34 +00:00
video_receive_stream.h
Rename FecConfig to UlpfecConfig in config.h.
2016-10-05 06:28:43 +00:00
video_send_stream.h
Rename FecConfig to UlpfecConfig in config.h.
2016-10-05 06:28:43 +00:00
voice_engine_configurations.h
Delete old video defines in engine config.
2016-10-07 05:07:36 +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.