Specifically, the tests that only use data channels shouldn't need any audio codec support; by using an audio decoder factory that supports no codecs, we ensure that this is the case. For completeness, I tried doing the same to the two tests that actually use audio and video; as expected, they fail, with messages like this: [000:032] (webrtcsession.cc:334): Failed to set remote sdp: Session error code: ERROR_CONTENT. Session error description: Failed to set local audio description recv parameters.. BUG=webrtc:5805 Review-Url: https://codereview.webrtc.org/2848563002 Cr-Commit-Position: refs/heads/master@{#17907}
Revert of CQ: Remove Linux ARM64 Debug trybot from default set. (patchset #1 id:1 of https://codereview.webrtc.org/2790263003/ )
Revert of Enable GN check for webrtc/base (patchset #9 id:350001 of https://codereview.webrtc.org/2840453004/ )
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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