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honghaiz 32f39968ce Re-apply change https://codereview.webrtc.org/1426673007/ 
Do not delete the turn port entry right away when the respective
connection is deleted. The dependency on asyncinvoker has been added
in chromium libjingle-nacl.

BUG=webrtc:5120

Review URL: https://codereview.webrtc.org/1450263002

Cr-Commit-Position: refs/heads/master@{#10679}
2015-11-17 19:36:37 +00:00
..
audio
Converted a bunch of error checking in Audio[Receive|Send]Stream to RTC_CHECKs instead. They should never fail.
2015-11-16 17:48:12 +00:00
base
Adding thread timeout for audio recorer thread in Java
2015-11-17 12:28:33 +00:00
build
Unconditionally build VP9 support.
2015-11-17 14:27:41 +00:00
call
Add UMA for send bwe and pacer bitrate.
2015-11-17 14:25:02 +00:00
common_audio
Changed queue implementation to the proposed vector-based solution.
2015-11-09 13:24:56 +00:00
common_video
common_video: rename interface -> include
2015-11-16 21:52:31 +00:00
examples
Add OpenSL ES enable setting to AppRTCDemo (part 2).
2015-11-17 18:12:46 +00:00
libjingle
cleanup: get rid of basicdefs.h include
2015-11-13 22:37:43 +00:00
modules
Add OpenSL ES enable setting to AppRTCDemo (part 2).
2015-11-17 18:12:46 +00:00
p2p
Re-apply change https://codereview.webrtc.org/1426673007/
2015-11-17 19:36:37 +00:00
sound
Convert usage of ARRAY_SIZE to arraysize.
2015-11-11 07:44:39 +00:00
system_wrappers
Revert of Create rtc::AtomicInt POD struct. (patchset #12 id:220001 of https://codereview.webrtc.org/1420043008/ )
2015-11-17 11:21:07 +00:00
test
common_video: rename interface -> include
2015-11-16 21:52:31 +00:00
tools
Remove webrtc/test/channel_transport/include
2015-11-11 09:34:35 +00:00
video
Add UMA for send bwe and pacer bitrate.
2015-11-17 14:25:02 +00:00
video_engine
Remove ViEEncoder::ScaleInputImage.
2015-11-17 14:52:59 +00:00
voice_engine
Remove include dirs from modules/{media_file,pacing}
2015-11-16 10:12:32 +00:00
.gitignore
Flesh out webrtc/.gitignore
2015-11-16 19:02:02 +00:00
audio_receive_stream.h
…
audio_send_stream.h
Move some send stream configuration into webrtc::AudioSendStream.
2015-11-16 15:34:59 +00:00
audio_state.h
Move VoiceEngineObserver into AudioSendStream so that we detect typing noises and return properly in GetStats().
2015-11-06 23:34:58 +00:00
BUILD.gn
Remove unused crypto.gni import.
2015-11-04 22:44:04 +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
…
common_types.h
Adding support for simulcast and spatial layers into VideoQualityTest
2015-11-02 15:23:24 +00:00
common.gyp
…
common.h
…
config.cc
…
config.h
Move some send stream configuration into webrtc::AudioSendStream.
2015-11-16 15:34:59 +00:00
engine_configurations.h
…
frame_callback.h
…
LICENSE
…
LICENSE_THIRD_PARTY
…
OWNERS
…
PATENTS
…
PRESUBMIT.py
…
README.chromium
…
rtc_unittests.isolate
…
stream.h
…
supplement.gypi
…
transport.h
…
typedefs.h
…
video_decoder.h
Add DecodedImageCallback::Decoded() function with custom decode time value.
2015-11-10 13:00:45 +00:00
video_encoder.h
Remove redudant encoder rate calls.
2015-10-29 15:30:29 +00:00
video_engine_tests.isolate
…
video_frame.h
common_video: rename interface -> include
2015-11-16 21:52:31 +00:00
video_receive_stream.h
…
video_renderer.h
…
video_send_stream.h
…
webrtc_examples.gyp
Remove webrtc/examples/android/media_demo.
2015-11-11 12:11:21 +00:00
webrtc_nonparallel_tests.isolate
Create isolate files for nonparallel tests.
2015-10-29 14:29:15 +00:00
webrtc_perf_tests.isolate
…
webrtc_tests.gypi
Move BitrateAllocator from BitrateController logic to Call.
2015-11-13 05:02:46 +00:00
webrtc.gyp
Move VoiceEngineObserver into AudioSendStream so that we detect typing noises and return properly in GetStats().
2015-11-06 23:34:58 +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.