Reason for revert: Speculative revert since all Android bots on WebRTC FYI started to fail when this CL landed. https://build.chromium.org/p/chromium.webrtc.fyi/builders/Android%20Tests%20%28dbg%29%20%28L%20Nexus6%29 Original issue's description: > If SRTP sessions exist, don't create new ones when applying answer. > > Instead, call the "Update" methods of SrtpSession, which will just call > srtp_update, instead of wiping out the session state completely. > > This was causing decryption to stop working when subsequent > offers/answers are applied. We don't know enough about SRTP to > understand the root cause, and I wasn't able to write an integration > test that reproduces the issue... But at least this fixes the bug that > can be reproduced reliably using Hangouts. > > BUG=webrtc:8251 > > Review-Url: https://codereview.webrtc.org/3019443002 > Cr-Commit-Position: refs/heads/master@{#19874} > Committed: https://webrtc.googlesource.com/src/+/5ada7acf603e90e71990e9d4ff8f49389f24958c TBR=zhihuang@webrtc.org,deadbeef@webrtc.org # Not skipping CQ checks because original CL landed more than 1 days ago. BUG=webrtc:8251 NOTRY=TRUE Review-Url: https://codereview.webrtc.org/3017543002 Cr-Commit-Position: refs/heads/master@{#19882}
Revert of If SRTP sessions exist, don't create new ones when applying answer. (patchset #1 id:1 of https://codereview.webrtc.org/3019443002/ )
Revert of Fix the video buffer size should take rtt into consideration (patchset #3 id:40001 of https://codereview.chromium.org/2980413002/ )
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 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://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
Description
The idea is to make CMake build for WebRTC m130 version - for audio processing module
Languages
C++
90.3%
Java
2.9%
C
2.2%
Objective-C++
2%
Python
1.3%
Other
1%