When remote side closes, opensslstreamadapter could return SR_EOS which will not trigger upper layer to clean up what's left in the StreamInterfaceChannel. The result of this is when there are more packets coming in, the Write on the StreamInterfaceChannel will overflow the buffer. The fix here is that when receiving the remote side close signal, we also close the underneath StreamInterfaceChannel which will clean up the queue to prevent overflow. BUG=574524 TBR=pthatcher@webrtc.org Review URL: https://codereview.webrtc.org/1566023002 Cr-Commit-Position: refs/heads/master@{#11751}
Revert of CQ: Disable linux_baremetal pending installation fix. (patchset #1 id:1 of https://codereview.webrtc.org/1710363002/ )
Reland of Add tools/mb to setup_links.py (patchset #1 id:1 of https://codereview.webrtc.org/1691723003/ )
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. This page is maintained by the Google Chrome team.
Development
See http://www.webrtc.org/native-code/development for instructions on how to get started developing with the native code.
More info
- Official web site: http://www.webrtc.org
- Master source code repo: https://chromium.googlesource.com/external/webrtc
- 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
Description
The idea is to make CMake build for WebRTC m130 version - for audio processing module
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