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platform-external-webrtc/webrtc
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ivoc 72c08edced Reenables several NetEq unittests on android. 
Several unittests were disabled on android, this CL will reenable them. One of
the tests was accidentally disabled on all platforms, and now no longer gives a
bitexact result.

BUG=webrtc:3343,webrtc:5349

Review URL: https://codereview.webrtc.org/1532903002

Cr-Commit-Position: refs/heads/master@{#11323}
2016-01-20 15:26:28 +00:00
..
api
Update API for Objective-C RTCMediaSource.
2016-01-11 22:39:05 +00:00
audio
Revert of Storing raw audio sink for default audio track. (patchset #7 id:120001 of https://codereview.chromium.org/1551813002/ )
2016-01-13 20:00:29 +00:00
base
Roll chromium_revision 7a4fb8d..f527e86 (370025:370073)
2016-01-19 12:47:24 +00:00
build
Make it possible to run peerconnection_unittests on Android.
2016-01-19 14:20:07 +00:00
call
Support REMB in combination with send-side BWE.
2016-01-20 15:14:03 +00:00
common_audio
Convert channel counts to size_t.
2016-01-13 00:26:55 +00:00
common_video
Switch to rtc::CriticalSection in IncomingVideoStream and remove one lock.
2016-01-20 12:38:00 +00:00
examples
Add a new interface for creating a udp socket in which it binds the socket to a network if the network handle is set.
2016-01-15 22:49:15 +00:00
libjingle
Remove DISABLED_ON_ macros.
2016-01-04 21:44:16 +00:00
modules
Reenables several NetEq unittests on android.
2016-01-20 15:26:28 +00:00
p2p
Remove two dead 'using' instances.
2016-01-18 23:24:36 +00:00
sound
OWNERS: Add * to .gyp{i,} everywhere.
2015-12-16 19:44:39 +00:00
system_wrappers
Remove ConditionVariableWrapper.
2016-01-20 12:36:42 +00:00
test
Reenables several NetEq unittests on android.
2016-01-20 15:26:28 +00:00
tools
Convert channel counts to size_t.
2016-01-13 00:26:55 +00:00
video
Support REMB in combination with send-side BWE.
2016-01-20 15:14:03 +00:00
voice_engine
Reland "Added option to specify a maximum file size when recording an AEC dump.", commit ae2c5ad12afc8cc29fe9c59dea432b697b871a87.
2016-01-15 11:06:41 +00:00
.gitignore
Flesh out webrtc/.gitignore
2015-11-16 19:02:02 +00:00
audio_receive_stream.h
Storing raw audio sink for default audio track.
2016-01-15 17:20:08 +00:00
audio_send_stream.h
Prepare the AudioSendStream to be hooked up to send-side BWE.
2015-12-07 09:26:32 +00:00
audio_state.h
Add comments about the Audio parts of the public Call API being WIP.
2015-12-03 12:06:31 +00:00
BUILD.gn
Re-land: "Use an explicit identifier in Config"
2016-01-14 12:32:51 +00:00
call.h
Move VoiceEngineObserver into AudioSendStream so that we detect typing noises and return properly in GetStats().
2015-11-06 23:34:58 +00:00
codereview.settings
Improve informative message in codereview.settings.
2015-11-10 10:08:36 +00:00
common_types.cc
Separating voice activity flag from audio level in RtpHeaderExtension.
2015-08-10 13:08:46 +00:00
common_types.h
Remove extra_options from VideoCodec.
2016-01-19 15:26:24 +00:00
common.gyp
…
common.h
Re-land: "Use an explicit identifier in Config"
2016-01-14 12:32:51 +00:00
config.cc
Wire up packet_id / send time callbacks to webrtc via libjingle.
2015-10-15 14:26:17 +00:00
config.h
Re-land: "Use an explicit identifier in Config"
2016-01-14 12:32:51 +00:00
engine_configurations.h
Remove VIDEOCODEC_* from engine_configurations.h.
2015-11-24 12:56:05 +00:00
frame_callback.h
Unify FrameType and VideoFrameType.
2015-10-19 09:39:15 +00:00
LICENSE
…
LICENSE_THIRD_PARTY
…
OWNERS
…
PATENTS
…
PRESUBMIT.py
…
README.chromium
…
rtc_unittests.isolate
…
stream.h
Wire up PacketTime to ReceiveStreams.
2015-09-08 12:36:23 +00:00
supplement.gypi
Roll chromium_revision 68cf0b8..aa8e58a (361406:361601)
2015-11-25 21:20:11 +00:00
transport.h
Reland of Merge webrtc/video_engine/ into webrtc/video/ (patchset #2 id:300001 of https://codereview.webrtc.org/1507903005/ )
2015-12-09 11:13:40 +00:00
typedefs.h
Update attributes to match gclibc's ansidecl.h
2016-01-08 12:51:47 +00:00
video_decoder.h
Expose codec implementation names in stats.
2015-12-18 15:01:23 +00:00
video_encoder.h
Expose codec implementation names in stats.
2015-12-18 15:01:23 +00:00
video_engine_tests.isolate
Add audio streams to CallTest and a first A/V call test.
2016-01-07 16:43:31 +00:00
video_frame.h
Add unit test for stand-alone denoiser and fixed some bugs.
2016-01-12 05:34:14 +00:00
video_receive_stream.h
Expose codec implementation names in stats.
2015-12-18 15:01:23 +00:00
video_renderer.h
Reland of Merge webrtc/video_engine/ into webrtc/video/ (patchset #2 id:300001 of https://codereview.webrtc.org/1507903005/ )
2015-12-09 11:13:40 +00:00
video_send_stream.h
Expose codec implementation names in stats.
2015-12-18 15:01:23 +00:00
webrtc_examples.gyp
Disable warnings failing when using Clang on Windows.
2015-12-16 22:05:36 +00:00
webrtc_nonparallel_tests.isolate
Create isolate files for nonparallel tests.
2015-10-29 14:29:15 +00:00
webrtc_perf_tests.isolate
Enable GoogleWifiTrace3Mbps simulations.
2015-05-04 12:26:26 +00:00
webrtc_tests.gypi
Remove extra_options from VideoCodec.
2016-01-19 15:26:24 +00:00
webrtc.gyp
Remove 'video_engine_core_unittests' binary.
2015-12-04 09:59:02 +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.