[BREAKGLASS] Append-only mirror of github.com/signalapp/webrtc
bd9e4a95eb
This CL is part of Optimized Scaling efforts. In Chromium, the native frame buffer is getting an optimized CropAndScale() implementation. To support HW accelerated scaling, returning pre-scaled images and skipping unnecessary intermediate downscales, WebRTC needs to 1) use CropAndScale instead of libyuv::XXXXScale and 2) only map buffers it actually intends to encode. - To achieve this, WebRTC encoders are updated to map kNative video buffers so that in a follow-up CL VideoStreamEncoder can stop mapping intermediate buffer sizes. In this CL LibvpxVp9Encoder is updated to map kNative buffers of pixel formats it supports and convert ToI420() if the kNative buffer is something else. A fake native buffer that keeps track of which resolutions were mapped, MappableNativeBuffer, is added. Because VP9 is currently an SVC encoder and not a simulcast encoder, it does not need to invoke CropAndScale. This CL also fixes MultiplexEncoderAdapter, but because it simply forwards frames it only cares about the pixel format when |supports_augmented_data_| is true so this is the only time we map it. Because this encoder is not used with kNative in practise, we don't care to make this path optimal. Bug: webrtc:12469, chromium:1157072 Change-Id: I74edf85b18eccd0d250776bbade7a6444478efce Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/212580 Commit-Queue: Henrik Boström <hbos@webrtc.org> Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org> Reviewed-by: Evan Shrubsole <eshr@google.com> Cr-Commit-Position: refs/heads/master@{#33526} |
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| api | ||
| audio | ||
| build_overrides | ||
| call | ||
| common_audio | ||
| common_video | ||
| data | ||
| docs | ||
| examples | ||
| g3doc | ||
| logging | ||
| media | ||
| modules | ||
| p2p | ||
| pc | ||
| resources | ||
| rtc_base | ||
| rtc_tools | ||
| sdk | ||
| stats | ||
| style-guide | ||
| system_wrappers | ||
| test | ||
| tools_webrtc | ||
| video | ||
| .clang-format | ||
| .git-blame-ignore-revs | ||
| .gitignore | ||
| .gn | ||
| .vpython | ||
| abseil-in-webrtc.md | ||
| AUTHORS | ||
| BUILD.gn | ||
| CODE_OF_CONDUCT.md | ||
| codereview.settings | ||
| DEPS | ||
| DIR_METADATA | ||
| ENG_REVIEW_OWNERS | ||
| LICENSE | ||
| license_template.txt | ||
| native-api.md | ||
| OWNERS | ||
| PATENTS | ||
| presubmit_test_mocks.py | ||
| presubmit_test.py | ||
| PRESUBMIT.py | ||
| pylintrc | ||
| README.chromium | ||
| README.md | ||
| style-guide.md | ||
| WATCHLISTS | ||
| webrtc_lib_link_test.cc | ||
| webrtc.gni | ||
| whitespace.txt | ||
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 here for instructions on how to get started developing with the native code.
Authoritative list of directories that contain the native API header files.
More info
- Official web site: http://www.webrtc.org
- Master source code repo: https://webrtc.googlesource.com/src
- 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
- Coding style guide
- Code of conduct
- Reporting bugs