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deadbeef 9794366ff0 Fixing memory leak in TurnServer. 
If the test TURN server received two allocate requests from the same
address, it was replacing the old allocation but not deleting it.

Also switching to std::unique_ptr to make it less likely for this to
pop up again.

Review-Url: https://codereview.webrtc.org/2114063002
Cr-Commit-Position: refs/heads/master@{#13449}
2016-07-12 18:04:57 +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
SurfaceViewRendererOnMeasureTest: Wait for frame size change to take effect
2016-07-12 14:48:22 +00:00
audio
Reland of "Move RtcEventLog object from inside VoiceEngine to Call.", "Fix to make the start/stop functions for the Rtc Eventlog non-virtual." and "Fix for RtcEventLog ObjC interface"
2016-07-04 14:07:03 +00:00
base
Handle initialization race in TaskQueue on Windows, discovered by drmemory.
2016-07-12 08:24:41 +00:00
build
Revert of Protect MessageQueue stop field with a critical section to avoid data races. (patchset #5 id:80001 of https://codereview.webrtc.org/2023193002/ )
2016-07-11 09:48:49 +00:00
call
Revert of Reland Issue 2061423003: Refactor NACK bitrate allocation (patchset #1 id:1 of https://codereview.webrtc.org/2131313002/ )
2016-07-08 18:02:02 +00:00
common_audio
Pull out the PostFilter to its own NonlinearBeamformer API
2016-07-02 00:19:32 +00:00
common_video
include planar functions for I420Rect()
2016-07-09 00:33:28 +00:00
examples
Camera2 in AppRTC Android Demo.
2016-07-06 07:58:35 +00:00
libjingle
GN: Update xmpp and p2p to match Chromium build
2016-06-18 11:23:08 +00:00
media
Revert of Clean up unused cricket::VideoCapturer inclusion from mediaengine.h (patchset #1 id:1 of https://codereview.webrtc.org/2147443002/ )
2016-07-12 12:54:26 +00:00
modules
Revert of Adds data logging in native AudioDeviceBuffer class (patchset #10 id:180001 of https://codereview.webrtc.org/2132613002/ )
2016-07-12 10:08:51 +00:00
p2p
Fixing memory leak in TurnServer.
2016-07-12 18:04:57 +00:00
pc
Reland of "Move RtcEventLog object from inside VoiceEngine to Call.", "Fix to make the start/stop functions for the Rtc Eventlog non-virtual." and "Fix for RtcEventLog ObjC interface"
2016-07-04 14:07:03 +00:00
sdk
Prefix bool variable with "should".
2016-07-11 22:09:32 +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
Add a fuzzer for the pseudoTCP implementation used by chromoting. This
2016-07-07 12:52:29 +00:00
tools
Revert of Visualization tool for WebrtcEventLogs (patchset #9 id:160001 of https://codereview.webrtc.org/1995523002/ )
2016-07-12 12:11:29 +00:00
video
Remove frame rate and target bitrate from ProtectionBitrateCalculator::SetEncodingData
2016-07-11 08:47:36 +00:00
voice_engine
Added voice_engine_unittests and voe_auto_test targets to GN.
2016-07-08 09:13:02 +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
Add AudioSendStream::SetMuted() method and use it in WVoMC::MuteStream().
2016-06-16 17:53:28 +00:00
audio_sink.h
…
audio_state.h
…
BUILD.gn
Revert of Visualization tool for WebrtcEventLogs (patchset #9 id:160001 of https://codereview.webrtc.org/1995523002/ )
2016-07-12 12:11:29 +00:00
call.h
Fix bug where min transmit bitrate wasn't working
2016-07-06 07:54:35 +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
Revert of Reland Issue 2061423003: Refactor NACK bitrate allocation (patchset #1 id:1 of https://codereview.webrtc.org/2131313002/ )
2016-07-08 18:02:02 +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
…
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
…
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
…
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
Revert of Reland Issue 2061423003: Refactor NACK bitrate allocation (patchset #1 id:1 of https://codereview.webrtc.org/2131313002/ )
2016-07-08 18:02:02 +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.