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1f3679884c
Start introducing ArrayView to AudioFrame and code that flows down from there. In this first step: * Add `data_view()` that returns a read-only ArrayView for the audio buffer. When AudioFrame is not initialized however, data_view() will return a nullptr whereas the current data() method never returns nullptr. * Add `mutable_data()` that requires two arguments for properly setting the samples per channel and number of channels that's required for accurately reserving the returned mutable ArrayView. A notable behavior change is that if the requested number of channels is larger than supported or the calculated buffer size is too large, the function will trigger a check. * Add TODOs for following work. Bug: chromium:335805780 Change-Id: I2937de800422589ebe6a3840b3caadf3d9ff8b00 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/347982 Reviewed-by: Per Åhgren <peah@webrtc.org> Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org> Cr-Commit-Position: refs/heads/main@{#42202} |
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| api | ||
| audio | ||
| build_overrides | ||
| call | ||
| common_audio | ||
| common_video | ||
| data | ||
| docs | ||
| examples | ||
| experiments | ||
| g3doc | ||
| infra | ||
| logging | ||
| media | ||
| modules | ||
| net/dcsctp | ||
| p2p | ||
| pc | ||
| resources | ||
| rtc_base | ||
| rtc_tools | ||
| sdk | ||
| stats | ||
| system_wrappers | ||
| test | ||
| tools_webrtc | ||
| video | ||
| .clang-format | ||
| .git-blame-ignore-revs | ||
| .gitignore | ||
| .gn | ||
| .mailmap | ||
| .style.yapf | ||
| .vpython3 | ||
| AUTHORS | ||
| BUILD.gn | ||
| CODE_OF_CONDUCT.md | ||
| codereview.settings | ||
| DEPS | ||
| DIR_METADATA | ||
| ENG_REVIEW_OWNERS | ||
| LICENSE | ||
| license_template.txt | ||
| native-api.md | ||
| OWNERS | ||
| OWNERS_INFRA | ||
| PATENTS | ||
| PRESUBMIT.py | ||
| presubmit_test.py | ||
| presubmit_test_mocks.py | ||
| pylintrc | ||
| pylintrc_old_style | ||
| README.chromium | ||
| README.md | ||
| WATCHLISTS | ||
| webrtc.gni | ||
| webrtc_lib_link_test.cc | ||
| 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: https://ci.chromium.org/p/webrtc/g/ci/console
- Coding style guide
- Code of conduct
- Reporting bugs
- Documentation