Only three items in the (rather large) header were actually used after InsertPacket: payloadType, timestamp and sequenceNumber. They are now put directly into Packet. This saves 129 bytes per Packet that no longer need to be allocated and deallocated. This also works towards decoupling NetEq from RTP. As part of that, I've moved the NACK code earlier in InsertPacketInternal, together with other things that directly reference the RTPHeader. BUG=webrtc:6549 Review-Url: https://codereview.webrtc.org/2411183003 Cr-Commit-Position: refs/heads/master@{#14658}
Revert of Only expose gflags target in non-Chromium and non-fuzzer builds. (patchset #1 id:40001 of https://codereview.webrtc.org/2321963002/ )
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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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