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webrtc_m130/webrtc
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henrik.lundin da8bbf6e3c NetEq: Change member variables for current RTP types to rtc::Optionals 
With this change, the value 0xFF is no longer used to flag that the RTP
type is unknown. Instead, an empty value for the rtc::Optional is used.

Review-Url: https://codereview.webrtc.org/2290153002
Cr-Commit-Position: refs/heads/master@{#13989}
2016-08-31 10:14:18 +00:00
..
androidjunit
Re-order and remove unused Java imports.
2016-08-02 13:44:19 +00:00
api
Reland of move the old AndroidVideoCapturer stack code. (patchset #1 id:1 of https://codereview.webrtc.org/2291583002/ )
2016-08-31 09:53:54 +00:00
audio
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
base
Do not build task_queue.cc and include from webrtc_overrides in Chrome.
2016-08-31 06:01:17 +00:00
build
GN: Use custom suppressions for lint in android.
2016-08-31 08:55:10 +00:00
call
Do not update stats for WebRTC.Call.EstimatedSendBitrateInKbps if we are not sending video.
2016-08-23 20:00:21 +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
Update AecDump API to match RtcEventLog API signature.
2016-08-30 19:58:14 +00:00
libjingle
GN: Update xmpp and p2p to match Chromium build
2016-06-18 11:23:08 +00:00
media
New static I420Buffer::Rotate method, to replace GetCopyWithRotationApplied.
2016-08-30 15:45:48 +00:00
modules
NetEq: Change member variables for current RTP types to rtc::Optionals
2016-08-31 10:14:18 +00:00
p2p
Add parameter to TransportController to not change ICE role on restart.
2016-08-27 03:59:35 +00:00
pc
Combining "SetTransportChannel" and "SetRtcpTransportChannel".
2016-08-27 04:42:20 +00:00
sdk
Update AecDump API to match RtcEventLog API signature.
2016-08-30 19:58:14 +00:00
stats
RTCStatsCollector and RTCPeerConnectionStats added.
2016-08-30 21:04:40 +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
Fix resource files for tools_unittests
2016-08-30 11:10:40 +00:00
video
Fixing TSan data race warning in video end-to-end tests.
2016-08-23 01:14:21 +00:00
voice_engine
Added a level indicator to new mixer.
2016-08-24 08:17:20 +00:00
.gitignore
…
audio_receive_stream.h
Add AudioReceiveStream::SetGain() method and use that in WVoMC::SetOutputVolume().
2016-06-17 15:30:58 +00:00
audio_send_stream.h
Variable audio bitrate.
2016-07-26 11:44:12 +00:00
audio_sink.h
Reland "Move webrtc/audio/audio_sink.h to webrtc/ and fix some dependencies."
2016-02-26 21:46:22 +00:00
audio_state.h
…
BUILD.gn
GN: Fix resource files for iOS test target
2016-08-30 09:53:54 +00:00
call.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
codereview.settings
Give a more specific URL for creating WebRTC checkout
2016-04-01 07:36:53 +00:00
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
Fix Chromium clang plugin warnings
2016-08-29 17:05:34 +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
Moved creation of AudioDecoderFactory to inside PeerConnectionFactory.
2016-06-13 14:35:01 +00:00
engine_configurations.h
Remove the VoEDtmf interface.
2016-03-08 12:11:00 +00:00
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
Add isolate files for Android tests
2016-04-18 03:08:28 +00:00
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
Replace scoped_ptr with unique_ptr in webrtc/common_video/
2016-03-02 11:41:40 +00:00
video_encoder.h
Add EncodedImageCallback::OnEncodedImage().
2016-08-03 00:46:47 +00:00
video_engine_tests_apk.isolate
Add isolate files for Android tests
2016-04-18 03:08:28 +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 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
Add isolate files for Android tests
2016-04-18 03:08:28 +00:00
webrtc_nonparallel_tests.isolate
…
webrtc_perf_tests_apk.isolate
Add isolate files for Android tests
2016-04-18 03:08:28 +00:00
webrtc_perf_tests.isolate
Add an HD resolution perf test.
2016-08-11 14:01:03 +00:00
webrtc_tests.gypi
GN: Fix test_support_unittests and MIPS compile issue.
2016-08-23 11:52:19 +00:00
webrtc.gyp
Remove webrtc_all target
2016-06-14 09:09:24 +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.