This reverts commit e30c27205148b34ba421184efe65f6a0780b436d (https://codereview.webrtc.org/1958053002/) Original reverted cl is in patch set #1. Changes in following patch sets. The cl now also make sure SendPacer starts with the configured bitrate provided in a call to CongestionController::SetBweBitrates)() It turns out that the failing tests in 609816 is due to a bug in the current code that runs the proper at 300kbit regardless of configured start bitrate. Original cl description: Remove SendPacer from ViEEncoder This CL moves the logic where the ViEEncoder pause if the pacer is full to the BitrateController. If the queue is full, the controller reports a bitrate of zero to Call (and BitrateAllocator) BUG=chromium:609816, webrtc:5687 TBR=mflodman@webrtc.org NOTRY=True // Due to bug in android_x86 cq builder.... Review-Url: https://codereview.webrtc.org/1958113003 Cr-Commit-Position: refs/heads/master@{#12688}
Revert of Remove webrtc/base/scoped_ptr.h (patchset #3 id:100001 of https://codereview.webrtc.org/1942823002/ )
Revert of Added webrtc/base/safe_conversions.h as a pseudonym (patchset #1 id:20001 of https://codereview.webrtc.org/1774933003/ )
Remove DropOldEvents from list of excluded tests. As of last week, the test no longer exists. (https://codereview.webrtc.org/1687703002/)
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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