A recent change (https://codereview.webrtc.org/2855143002/) introduced a bug in RtcEventLogSource::NextPacket(). The rtp_packet_index_ must be incremented when a valid packet is found and delivered. Otherwise, the same packet will be delivered over and over again. The recent change also altered the way that audio packets are sifted out. Now, the RTP header is always parsed before discarding any non-audio packets. This means that RtpHeaderParser::Parse is always called, also with video packets, which sometimes contain padding. When header-only dumps (such as RtcEventLogs) are created, the payload is stripped, and the payload length is equal to the RTP header length. However, if the original packet was padded, the RTP header will carry information about this padding length, and the parser will check that the pyaload length is at least the header + padding. This is not the case for header-only dumps, and the parser will return an error. In this CL, we ignore that error when a header-only packet has padding length larger than 0. BUG=webrtc:7538 Review-Url: https://codereview.webrtc.org/2912323003 Cr-Commit-Position: refs/heads/master@{#18385}
Reland of Split iOS sdk in to separate targets (patchset #1 id:1 of https://codereview.webrtc.org/2911053002/ )
Reland of Enabling
gn check on webrtc/test (patchset #1 id:1 of https://codereview.webrtc.org/2920763002/ )
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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