Reason for revert: Fix RtpStreamReceiver to not recover RTX packets with incorrect SSRC. Original issue's description: > Revert of Only compare sequence numbers from the same SSRC in ForwardErrorCorrection. (patchset #5 id:120001 of https://codereview.webrtc.org/2893293003/ ) > > Reason for revert: > Breaks fuzzer. > > Original issue's description: > > Only compare sequence numbers from the same SSRC in ForwardErrorCorrection. > > > > Prior to this CL, the ForwardErrorCorrection state would be reset whenever > > the difference in sequence numbers of the last recovered media packet > > and the new packet (media or FEC) was too large. This comparison did not > > take into account that FlexFEC uses a different SSRC for the FEC packets, > > meaning that the the state would be reset very frequently when FlexFEC > > is used. This should not have led to any major problems, except for a > > decreased decoding efficiency. > > > > This CL verifies that whenever we compare sequence numbers in > > ForwardErrorCorrection, they do indeed belong to the same SSRC. > > > > BUG=webrtc:5654 > > > > Review-Url: https://codereview.webrtc.org/2893293003 > > Cr-Commit-Position: refs/heads/master@{#18399} > > Committed:1476a9d789> > TBR=stefan@webrtc.org,holmer@google.com > # Not skipping CQ checks because original CL landed more than 1 days ago. > BUG=webrtc:5654 > > Review-Url: https://codereview.webrtc.org/2919313005 > Cr-Commit-Position: refs/heads/master@{#18446} > Committed:92732ecc5cR=stefan@webrtc.org BUG=webrtc:5654 Review-Url: https://codereview.webrtc.org/2918333002 Cr-Commit-Position: refs/heads/master@{#18827}
Reland of Only compare sequence numbers from the same SSRC in ForwardErrorCorrection. (patchset #1 id:1 of https://codereview.webrtc.org/2919313005/ )
Reland of Moving webrtc.gni up one level from build/ (patchset #1 id:1 of https://codereview.webrtc.org/2657563002/ )
Reland of PyLint fixes for tools-webrtc and webrtc/tools (patchset #1 id:1 of https://codereview.webrtc.org/2737233003/ )
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://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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