commit | 64e16080b12b3f79550f3be1db70e4bafb51cf06 | [log] [tgz] |
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author | Philip Eliasson <philipel@webrtc.org> | Thu Jun 05 08:54:35 2025 |
committer | WebRTC LUCI CQ <webrtc-scoped@luci-project-accounts.iam.gserviceaccount.com> | Thu Jun 05 10:39:21 2025 |
tree | 67cdafda79387e06b44da3173ef98c60441b33f7 | |
parent | 958c5f2876dd481692a1768bd0d7f5968095a708 [diff] |
Reland "Add support for setting CSRCs on audio and video senders" This reverts commit 84f48e824a68e7dc72b7ed229726341e12271157. Reason for revert: revert breaks ToT Bug: b/410811496 Original change's description: > Revert "Add support for setting CSRCs on audio and video senders" > > This reverts commit dd3768ef7266e0e4840e883a0f652e3c75887cad. > > Reason for revert: breaks downstream projects > > Bug: b/410811496 > Original change's description: > > Add support for setting CSRCs on audio and video senders > > > > With this change, CSRCs can be added to video packets sent via > > RTPSenderVideo::SendEncodedImage. This is implemented by keeping a list > > of CSRCs in the calling class RtpVideoSender, which is included in all > > calls to SendEncodedImage. > > > > This CL is part of a chain, with the next being > > https://webrtc-review.googlesource.com/c/src/+/392961. Ultimately, the > > point is to support setting the CSRC list via RtpEncodingParameters. > > This is done in https://webrtc-review.googlesource.com/c/src/+/392980. > > > > Bug: b/410811496 > > Change-Id: I2b9c430c6b19b423f2f29cf8e81b04ad04c2b915 > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/392940 > > Reviewed-by: Harald Alvestrand <hta@webrtc.org> > > Reviewed-by: Jonas Oreland <jonaso@webrtc.org> > > Commit-Queue: Helmer Nylén <helmern@google.com> > > Cr-Commit-Position: refs/heads/main@{#44824} > > Bug: b/410811496 > No-Presubmit: true > No-Tree-Checks: true > No-Try: true > Change-Id: I0c7730b89468740e2692dd52eb72fcd67cf0f040 > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/395160 > Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com> > Owners-Override: Philip Eliasson <philipel@webrtc.org> > Commit-Queue: Philip Eliasson <philipel@webrtc.org> > Cr-Commit-Position: refs/heads/main@{#44833} Bug: b/410811496 Change-Id: I9fab185d8ab6e9dfe7583f409f7d6dd14fc4e429 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/395002 Owners-Override: Philip Eliasson <philipel@webrtc.org> Commit-Queue: Philip Eliasson <philipel@webrtc.org> Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org> Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com> Cr-Commit-Position: refs/heads/main@{#44835}
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.
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.