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platform-external-webrtc/webrtc
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henrik.lundin ebd9fd7918 NetEq: Add a few EXPECT_CALLs to PacketBuffer unittests 
This will silence a few noisy print-outs.

Review-Url: https://codereview.webrtc.org/2290173002
Cr-Commit-Position: refs/heads/master@{#14005}
2016-08-31 20:03:40 +00:00
..
androidjunit
Re-order and remove unused Java imports.
2016-08-02 13:44:19 +00:00
api
Add signaling to support ICE renomination.
2016-08-31 15:18:22 +00:00
audio
Move webrtc/audio_*.h to webrtc/api/call
2016-08-31 14:33:14 +00:00
base
Added an empty() method to rtc::Buffer
2016-08-31 15:40:08 +00:00
build
Emit landmine for Android builds to solve compile lint errors
2016-08-31 15:24:06 +00:00
call
Move webrtc/audio_*.h to webrtc/api/call
2016-08-31 14:33:14 +00:00
common_audio
GN: Fix common_audio linking on iOS
2016-08-30 11:17:18 +00:00
common_video
New static I420Buffer::Rotate method, to replace GetCopyWithRotationApplied.
2016-08-30 15:45:48 +00:00
examples
Add local capturer on mac
2016-08-31 17:23:31 +00:00
libjingle
GN: Update xmpp and p2p to match Chromium build
2016-06-18 11:23:08 +00:00
media
Move webrtc/audio_*.h to webrtc/api/call
2016-08-31 14:33:14 +00:00
modules
NetEq: Add a few EXPECT_CALLs to PacketBuffer unittests
2016-08-31 20:03:40 +00:00
p2p
Add signaling to support ICE renomination.
2016-08-31 15:18:22 +00:00
pc
Add signaling to support ICE renomination.
2016-08-31 15:18:22 +00:00
sdk
Update AecDump API to match RtcEventLog API signature.
2016-08-30 19:58:14 +00:00
stats
RTCStatsCollector: timestamps updated.
2016-08-31 14:57:44 +00:00
system_wrappers
Remove Chromium Clang warnings only on Windows.
2016-08-29 09:56:10 +00:00
test
GN: Fix resource files for iOS test target
2016-08-30 09:53:54 +00:00
tools
Move webrtc/audio_*.h to webrtc/api/call
2016-08-31 14:33:14 +00:00
video
Fixing TSan data race warning in video end-to-end tests.
2016-08-23 01:14:21 +00:00
voice_engine
Move webrtc/audio_*.h to webrtc/api/call
2016-08-31 14:33:14 +00:00
.gitignore
…
BUILD.gn
Use max optimization when compiling neteq_performance_test.
2016-08-31 15:22:13 +00:00
call.h
Move webrtc/audio_*.h to webrtc/api/call
2016-08-31 14:33:14 +00:00
codereview.settings
…
common_types.cc
FileWrapper[Impl] modifications and actually remove the "Impl" class.
2016-06-15 17:30:18 +00:00
common_types.h
Reland of actor NACK bitrate allocation (patchset #1 id:1 of https://codereview.webrtc.org/2131913003/ )
2016-07-13 16:11:38 +00:00
common.cc
Fix Chromium clang plugin warnings
2016-08-29 17:05:34 +00:00
common.gyp
Move webrtc/audio_*.h to webrtc/api/call
2016-08-31 14:33:14 +00:00
common.h
Fix Chromium clang plugin warnings
2016-08-29 17:05:34 +00:00
config.cc
Avoid unnecessary HW video encoder reconfiguration
2016-06-16 19:08:11 +00:00
config.h
Revert of Add task queue to Call. (patchset #42 id:840001 of https://codereview.webrtc.org/2060403002/ )
2016-08-16 09:40:59 +00:00
DEPS
Move webrtc/audio_*.h to webrtc/api/call
2016-08-31 14:33:14 +00:00
engine_configurations.h
…
LICENSE
…
LICENSE_THIRD_PARTY
…
OWNERS
Add stefan@ to webrtc/OWNERS.
2016-08-10 15:49:59 +00:00
PATENTS
…
PRESUBMIT.py
…
README.chromium
…
rtc_unittests_apk.isolate
…
rtc_unittests.isolate
…
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 EncodedImageCallback::OnEncodedImage().
2016-08-03 00:46:47 +00:00
video_engine_tests_apk.isolate
…
video_engine_tests.isolate
…
video_frame.h
Add TODO comments on deprecated VideoFrame methods.
2016-06-22 15:47:52 +00:00
video_receive_stream.h
Add periodic logging of video stats.
2016-08-11 15:41:26 +00:00
video_send_stream.h
Revert of Add task queue to Call. (patchset #42 id:840001 of https://codereview.webrtc.org/2060403002/ )
2016-08-16 09:40:59 +00:00
webrtc_examples.gyp
Add GN targets for AppRTC Demo on Android.
2016-06-29 12:55:12 +00:00
webrtc_nonparallel_tests_apk.isolate
…
webrtc_nonparallel_tests.isolate
…
webrtc_perf_tests_apk.isolate
…
webrtc_perf_tests.isolate
Add an HD resolution perf test.
2016-08-11 14:01:03 +00:00
webrtc_tests.gypi
Move webrtc/audio_*.h to webrtc/api/call
2016-08-31 14:33:14 +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.