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webrtc_m130/webrtc
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deadbeef f15fb452ef Fix occasional race in VideoCapturerTrackSource seen by memcheck bot. 
The issue was that VideoCapturerTrackSource was adding a reference
to itself, causing it to not be deleted even after no external objects
reference it. The objects underneath it (threads for instance) may
then be destroyed before the object dereferences them.

BUG=webrtc:6487

Review-Url: https://codereview.webrtc.org/2717023002
Cr-Commit-Position: refs/heads/master@{#16841}
2017-02-25 21:37:59 +00:00
..
api
Move rtc_api_unittests into rtc_unittests.
2017-02-24 09:17:43 +00:00
audio
Replace AudioReceiveStream::DeliverRtp with OnRtpPacket.
2017-02-21 14:28:10 +00:00
base
Adding rtc::Unretained to allow avoiding rtc::Bind reference capture.
2017-02-25 20:56:20 +00:00
call
Introduce a new constructor to PlatformThread.
2017-02-22 19:22:05 +00:00
common_audio
Fix for left shift of potentially negative value.
2017-02-09 11:05:59 +00:00
common_video
Introduce a new constructor to PlatformThread.
2017-02-22 19:22:05 +00:00
examples
Reland of move some warning suppressions from SocketRocket. (patchset #2 id:60001 of https://codereview.webrtc.org/2718703002/ )
2017-02-24 16:09:18 +00:00
logging
Introduce a new constructor to PlatformThread.
2017-02-22 19:22:05 +00:00
media
Fix issue where video scaling gets stuck at low resolution
2017-02-22 17:30:27 +00:00
modules
Fix busy loop in FakeAudioDeviceModule.
2017-02-25 17:05:22 +00:00
ortc
Adding placeholder ortc_unittests target.
2017-02-23 07:45:38 +00:00
p2p
Accept SDP with TRANSPORT attributes missing from bundled m= sections.
2017-02-23 03:35:18 +00:00
pc
Fix occasional race in VideoCapturerTrackSource seen by memcheck bot.
2017-02-25 21:37:59 +00:00
sdk
Add the OnAddTrack callback for Objective-C wrapper.
2017-02-24 20:50:48 +00:00
stats
RTCRTPStreamStats.mediaTrackId renamed to trackId.
2017-01-27 14:35:16 +00:00
system_wrappers
Introduce a new constructor to PlatformThread.
2017-02-22 19:22:05 +00:00
test
Reland of Add optional visualization file writers to VideoProcessor tests. (patchset #1 id:1 of https://codereview.webrtc.org/2708103002/ )
2017-02-22 09:26:59 +00:00
tools
Propagate packet pacing information to SendTimeHistory.
2017-02-23 10:56:13 +00:00
video
Revert of Set scaling limit at 320 * 180 for all implementations. (patchset #2 id:20001 of https://codereview.webrtc.org/2709153002/ )
2017-02-25 19:30:55 +00:00
voice_engine
Propagate packet pacing information to SendTimeHistory.
2017-02-23 10:56:13 +00:00
.gitignore
…
BUILD.gn
Removes a test suite that is no longer used or maintained.
2017-02-24 11:36:20 +00:00
call.h
Moved call.h and most of api/call/* into call/
2016-12-07 12:53:04 +00:00
codereview.settings
…
common_types.cc
Reject XR TargetBitrate items with unsupported layer indices
2016-12-06 14:09:00 +00:00
common_types.h
Remove the Windows Wave audio device implementation.
2017-02-17 22:48:07 +00:00
config.cc
Remove FlexfecConfig and replace with specific struct in VideoSendStream.
2017-01-16 14:59:19 +00:00
config.h
Remove FlexfecConfig and replace with specific struct in VideoSendStream.
2017-01-16 14:59:19 +00:00
DEPS
Move VideoFrame and related declarations to webrtc/api/video.
2017-01-10 15:44:26 +00:00
LICENSE
…
LICENSE_THIRD_PARTY
Delete unused spreadsort implementation.
2016-12-05 11:03:26 +00:00
no_op_function.cc
Moving no_op_function.cc out of webrtc/build
2017-01-24 13:25:12 +00:00
OWNERS
Cleanup unused rules in webrtc/DEPS + add kjellander to OWNERS for it
2016-11-29 10:52:22 +00:00
PATENTS
…
PRESUBMIT.py
…
README.chromium
…
typedefs.h
Only define NO_RETURN if undefined
2017-01-30 08:54:19 +00:00
video_decoder.h
Reland of Add QP sum stats for received streams. (patchset #2 id:300001 of https://codereview.webrtc.org/2680893002/ )
2017-02-09 12:53:45 +00:00
video_encoder.h
Move usage of QualityScaler to ViEEncoder.
2016-11-29 09:44:22 +00:00
video_frame.h
Move VideoFrame and related declarations to webrtc/api/video.
2017-01-10 15:44:26 +00:00
video_receive_stream.h
Reland of Add QP sum stats for received streams. (patchset #2 id:300001 of https://codereview.webrtc.org/2680893002/ )
2017-02-09 12:53:45 +00:00
video_send_stream.h
Remove FlexfecConfig and replace with specific struct in VideoSendStream.
2017-01-16 14:59:19 +00:00
webrtc.gni
Move iOS tests to XCTest from gtest.
2017-02-14 12:58:56 +00:00
whitespace.txt
Moving whitespace file up by one folder
2017-01-24 13:19:42 +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.