Reason for revert: Oh dear, this broke compilation. I guess more was built on top of this CL before I reverted it. Reverting now for futher investigation (and re-land using CQ) Original issue's description: > Revert of Add aecdump support to audioproc_f. (patchset #8 id:200001 of https://codereview.webrtc.org/1409943002/ ) > > Reason for revert: > This breaks iOS GYP generation as described on http://www.webrtc.org/native-code/ios > I'm going to drive getting the build_with_libjingle=1 setting removed from the bots to match the official instructions. > > See https://code.google.com/p/webrtc/issues/detail?id=4653 for more context, as this is exactly what that issue tries to solve. > > Original issue's description: > > Add aecdump support to audioproc_f. > > > > Add a new interface to abstract away file operations. This CL temporarily > > removes support for dumping the output of reverse streams. It will be easy to > > restore in the new framework, although we may decide to only allow it with > > the aecdump format. > > > > We also now require the user to specify the output format, rather than > > defaulting to the input format. > > > > TEST=Bit-exact output to the previous audioproc_f version using an input wav > > file, and to the legacy audioproc using an aecdump file. > > > > Committed: https://crrev.com/bdafe31b86e9819b0adb9041f87e6194b7422b08 > > Cr-Commit-Position: refs/heads/master@{#10460} > > TBR=aluebs@webrtc.org,peah@webrtc.org,andrew@webrtc.org > NOPRESUBMIT=true > NOTREECHECKS=true > NOTRY=true > > Committed: https://crrev.com/d279941bb54bfdc6e7324bf36cac76581474b96d > Cr-Commit-Position: refs/heads/master@{#10523} TBR=aluebs@webrtc.org,peah@webrtc.org,andrew@webrtc.org NOPRESUBMIT=true NOTREECHECKS=true NOTRY=true Review URL: https://codereview.webrtc.org/1419953010 Cr-Commit-Position: refs/heads/master@{#10524}
Revert of Exclude offline bots from CQ config. (patchset #1 id:1 of https://codereview.webrtc.org/1420283013/ )
Reland of Add aecdump support to audioproc_f. (patchset #2 id:250001 of https://codereview.webrtc.org/1423693008/ )
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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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