[BREAKGLASS] Append-only mirror of github.com/signalapp/webrtc
e448a3fb54
Call DataChannelObserver::OnBufferedAmountChange on each successful send. Previously, the observer would get notified of buffered amount changes only when queued send data is consumed. Data gets queued only if it cannot be sent right away. According to the WebRTC standard[1], bufferedamount should be increased before each sent and decreased after each successful sent. Update implementation to be standard compliant. Design doc: http://doc/1lorHBn-GMn5U0T0RQANxrsW0pXhw8XGZM-xZyVUOW90 [1] https://w3c.github.io/webrtc-pc/#dom-datachannel-bufferedamount Bug: chromium:878682 Change-Id: Ife009d30c4a18dced9a54cf600a445bb1f02561d Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/123237 Commit-Queue: Marina Ciocea <marinaciocea@webrtc.org> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Reviewed-by: Henrik Boström <hbos@webrtc.org> Cr-Commit-Position: refs/heads/master@{#27057} |
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| api | ||
| audio | ||
| build_overrides | ||
| call | ||
| common_audio | ||
| common_video | ||
| crypto | ||
| data | ||
| examples | ||
| 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 | ||
| common_types.h | ||
| DEPS | ||
| 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.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 http://www.webrtc.org/native-code/development 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