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Honghai Zhang d8f6fc4656 Make the state transition for a PortAllocatorSession in each derived class instead of in the base class. 
Putting them in the base class may potentially break subclasses if they
have not called the same method in the base class.

R=deadbeef@webrtc.org, pthatcher@webrtc.org

Review URL: https://codereview.webrtc.org/2120733002 .

Cr-Commit-Position: refs/heads/master@{#13372}
2016-07-02 00:31:24 +00:00
..
androidjunit
Reland of Android: Change camera fps range selection (patchset #1 id:1 of https://codereview.webrtc.org/2021233002/ )
2016-06-01 07:44:07 +00:00
api
Add pruneTurnPorts to the java RTCConfiguration.
2016-07-01 21:40:53 +00:00
audio
Revert of Move RtcEventLog object from inside VoiceEngine to Call. (patchset #16 id:420001 of https://codereview.webrtc.org/1748403002/ )
2016-06-30 07:59:49 +00:00
base
If continual gathering is enabled,
2016-07-01 20:59:39 +00:00
build
Remove dead GYP target audio_device_module_java
2016-06-20 08:43:42 +00:00
call
Auto pause video streams based on encoder target bitrate.
2016-07-01 11:04:10 +00:00
common_audio
Pull out the PostFilter to its own NonlinearBeamformer API
2016-07-02 00:19:32 +00:00
common_video
Add a race-checking mechanism.
2016-07-01 10:45:29 +00:00
examples
Dispose audio source correctly in AppRTC Demo on Android.
2016-06-30 14:57:22 +00:00
libjingle
GN: Update xmpp and p2p to match Chromium build
2016-06-18 11:23:08 +00:00
media
Revert of Workaround for clang bug http://llvm.org/PR28348. (patchset #1 id:1 of https://codereview.webrtc.org/2110043003/ )
2016-06-30 09:32:37 +00:00
modules
Pull out the PostFilter to its own NonlinearBeamformer API
2016-07-02 00:19:32 +00:00
p2p
Make the state transition for a PortAllocatorSession in each derived class instead of in the base class.
2016-07-02 00:31:24 +00:00
pc
Revert of Move RtcEventLog object from inside VoiceEngine to Call. (patchset #16 id:420001 of https://codereview.webrtc.org/1748403002/ )
2016-06-30 07:59:49 +00:00
sdk
Add pruneTurnPorts to the IOS RTCConfiguration.
2016-07-01 21:22:30 +00:00
system_wrappers
UMA log for audio_device Init and Start(Playout|Recording). Make Init return a more specific error code, if possible.
2016-07-01 11:35:31 +00:00
test
Revert of Move RtcEventLog object from inside VoiceEngine to Call. (patchset #16 id:420001 of https://codereview.webrtc.org/1748403002/ )
2016-06-30 07:59:49 +00:00
tools
New rtc dump analyzing tool in Python
2016-06-20 14:39:21 +00:00
video
Auto pause video streams based on encoder target bitrate.
2016-07-01 11:04:10 +00:00
voice_engine
Revert of Move RtcEventLog object from inside VoiceEngine to Call. (patchset #16 id:420001 of https://codereview.webrtc.org/1748403002/ )
2016-06-30 07:59:49 +00:00
.gitignore
Flesh out webrtc/.gitignore
2015-11-16 19:02:02 +00:00
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
Add AudioSendStream::SetMuted() method and use it in WVoMC::MuteStream().
2016-06-16 17:53:28 +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
Add comments about the Audio parts of the public Call API being WIP.
2015-12-03 12:06:31 +00:00
BUILD.gn
Build webrtc_nonparallel_tests under GN.
2016-06-27 17:31:56 +00:00
call.h
Revert of Move RtcEventLog object from inside VoiceEngine to Call. (patchset #16 id:420001 of https://codereview.webrtc.org/1748403002/ )
2016-06-30 07:59:49 +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
Avoid unnecessary HW video encoder reconfiguration
2016-06-16 19:08:11 +00:00
common.gyp
Revert of Add missing headers and fix some missing dependencies (patchset #1 id:20001 of https://codereview.webrtc.org/1990593002/ )
2016-05-18 12:00:55 +00:00
common.h
New module for the adaptive level controlling functionality in the audio processing module
2016-06-29 22:26:19 +00:00
config.cc
Avoid unnecessary HW video encoder reconfiguration
2016-06-16 19:08:11 +00:00
config.h
Avoid unnecessary HW video encoder reconfiguration
2016-06-16 19:08:11 +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 kwiberg@webrtc.org as root owner.
2016-06-14 09:39:40 +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
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
[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
Remove QualityScaler framerate reduction.
2016-04-26 11:37:20 +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
Movable support for VideoReceiveStream::Config and avoid copies.
2016-06-10 15:58:12 +00:00
video_send_stream.h
Remove webrtc/stream.h and unutilized inheritance.
2016-05-02 03:18:36 +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
Create isolate files for nonparallel tests.
2015-10-29 14:29:15 +00:00
webrtc_perf_tests_apk.isolate
Add isolate files for Android tests
2016-04-18 03:08:28 +00:00
webrtc_perf_tests.isolate
Fix missing resource file in webrtc_perf_tests.isolate
2016-06-17 07:39:49 +00:00
webrtc_tests.gypi
New module for the adaptive level controlling functionality in the audio processing module
2016-06-29 22:26:19 +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.