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7b69a44c8b
Original patch contributed by andrey.semashev@gmail.com. In PhysicalSocketServer::WaitEpoll(), the loop verifies that the signalled dispatcher is in dispatchers_ set. It does so by looking up the dispatcher pointer in the set. This is vulnerable to the ABA problem because one dispatcher may be removed and destroyed and another created and added with the same address before epoll reports an event for the old dispatcher. The same issue exists for other Wait implementations, if a dispatcher is removed and a new one added with the same socket handle is the old. This is avoided by using a 64-bit key for looking up the dispatcher in the set. The key is set from a running counter which gets incremented when a dispatcher is added to the set, so even if the same dispatcher pointer is added, removed and added again, the key value will be different. This changes the storage of dispatchers_ from a set to a flat_hash_map, which uses a bit more memory but has faster lookup (O(1) as opposed to O(log n)). Bug: webrtc:11124 Change-Id: I6d206e1a367b58ba971edca9b48af7664384b797 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/181027 Commit-Queue: Taylor <deadbeef@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#32019} |
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| call | ||
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| common_video | ||
| data | ||
| docs | ||
| examples | ||
| logging | ||
| media | ||
| modules | ||
| p2p | ||
| pc | ||
| resources | ||
| rtc_base | ||
| rtc_tools | ||
| sdk | ||
| stats | ||
| style-guide | ||
| system_wrappers | ||
| test | ||
| tools_webrtc | ||
| video | ||
| .clang-format | ||
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| abseil-in-webrtc.md | ||
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| BUILD.gn | ||
| CODE_OF_CONDUCT.md | ||
| codereview.settings | ||
| common_types.h | ||
| DEPS | ||
| ENG_REVIEW_OWNERS | ||
| LICENSE | ||
| license_template.txt | ||
| native-api.md | ||
| OWNERS | ||
| PATENTS | ||
| PRESUBMIT.py | ||
| presubmit_test.py | ||
| presubmit_test_mocks.py | ||
| pylintrc | ||
| README.chromium | ||
| README.md | ||
| style-guide.md | ||
| WATCHLISTS | ||
| webrtc.gni | ||
| webrtc_lib_link_test.cc | ||
| whitespace.txt | ||
WebRTC is a free, open software project that provides browsers and mobile applications with Real-Time Communications (RTC) capabilities via simple APIs. The WebRTC components have been optimized to best serve this purpose.
Our mission: To enable rich, high-quality RTC applications to be developed for the browser, mobile platforms, and IoT devices, and allow them all to communicate via a common set of protocols.
The WebRTC initiative is a project supported by Google, Mozilla and Opera, amongst others.
Development
See here for instructions on how to get started developing with the native code.
Authoritative list of directories that contain the native API header files.
More info
- Official web site: http://www.webrtc.org
- Master source code repo: https://webrtc.googlesource.com/src
- Samples and reference apps: https://github.com/webrtc
- Mailing list: http://groups.google.com/group/discuss-webrtc
- Continuous build: http://build.chromium.org/p/client.webrtc
- Coding style guide
- Code of conduct