Reason for revert: Reland after fixing broken perf tests. Original issue's description: > Revert of Set scaling limit at 320 * 180 for all implementations. (patchset #2 id:20001 of https://codereview.webrtc.org/2709153002/ ) > > Reason for revert: > Looks like webrtc_perf_test started failing on linux, mac and windows after this cl landed. > > Example failure: > > https://build.chromium.org/p/client.webrtc.perf/builders/Linux%20Trusty/builds/1386/steps/webrtc_perf_tests/logs/stdio > > [ RUN ] CallPerfTest.ReceivesCpuOveruseAndUnderuse > ../../webrtc/call/call_perf_tests.cc:522: Failure > Value of: Wait() > Actual: false > Expected: true > Timed out before receiving an overuse callback. > [ FAILED ] CallPerfTest.ReceivesCpuOveruseAndUnderuse (120056 ms) > > > Original issue's description: > > Set scaling limit at 320 * 180 for all implementations. > > > > The MediaCodec decoder on android has trouble decoding video at > > so low resolutions. We set the limit a bit higher for all implementations > > pending a robust software fallback implementation for MediaCodec. > > > > BUG=webrtc:7206 > > > > Review-Url: https://codereview.webrtc.org/2709153002 > > Cr-Commit-Position: refs/heads/master@{#16798} > > Committed:560ddb7321> > TBR=magjed@webrtc.org,sprang@webrtc.org,kthelgason@webrtc.org > # Not skipping CQ checks because original CL landed more than 1 days ago. > BUG=webrtc:7206 > > Review-Url: https://codereview.webrtc.org/2711913007 > Cr-Commit-Position: refs/heads/master@{#16839} > Committed:37510bf094TBR=magjed@webrtc.org,sprang@webrtc.org,tommi@webrtc.org # Not skipping CQ checks because original CL landed more than 1 days ago. BUG=webrtc:7206 Review-Url: https://codereview.webrtc.org/2718013002 Cr-Commit-Position: refs/heads/master@{#16853}
Reland of Set scaling limit at 320 * 180 for all implementations. (patchset #1 id:1 of https://codereview.webrtc.org/2711913007/ )
Reland of Moving webrtc.gni up one level from build/ (patchset #1 id:1 of https://codereview.webrtc.org/2657563002/ )
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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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