commit | 79e5e721b54b3821f33be96bda3f1be69429cd53 | [log] [tgz] |
---|---|---|
author | Henrik Boström <hbos@webrtc.org> | Wed Jan 22 14:13:37 2025 |
committer | WebRTC LUCI CQ <webrtc-scoped@luci-project-accounts.iam.gserviceaccount.com> | Wed Jan 22 16:27:25 2025 |
tree | d4c8cf211e245210a73181025375ef53c9872efa | |
parent | 49ac6b758cc3c28be2fc13028a829f016b453d39 [diff] |
Add unidirectional codec support ("offer to send" use case). This CL implements allowing sendonly codecs in setCodecPreferences(), i.e. this spec PR: https://github.com/w3c/webrtc-pc/pull/3018. It also makes the setCodecPreferences() ignore level IDs in the filtering algorithm (but not in the sCP method call) as per this spec PR: https://github.com/w3c/webrtc-pc/pull/3023. In short, before this CL, setCodecPreferences() threw an exception if a codec was preferred that is not present in receiver codec capabilities. After this CL, setCodecPreferences() allows you to prefer codecs that are *either* in the sender capabilities *or* the receiver capabilities. - This allows you to "offer to send", i.e. prefer sendonly codecs on a sendonly transceiver. - The filtering on direction is handled by RtpTransceiver::filtered_codec_preferences() which is called during SDP offer/answer (sdp_offer_answer.cc). Also as per spec changes, if this filtering results in not having any codecs to offer or answer then this results in not having any codec preferences as opposed to throwing an exception (old behavior). - Two old peer_connection_media_unittest.cc tests are updated to reflect the API failing less. This CL adds both unit tests (rtp_transceiver_unittest.cc) and full stack integration tests (peer_connection_encodings_integrationtest.cc). It also makes us pass the following Web Platform Tests in Chrome: https://source.chromium.org/chromium/chromium/src/+/main:third_party/blink/web_tests/external/wpt/webrtc/protocol/h265-level-id.https.html Bug: chromium:381407888 Change-Id: I98a5ad1acccb56db0538e4d47975b8a725102c33 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/374520 Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org> Commit-Queue: Henrik Boström <hbos@webrtc.org> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Reviewed-by: Evan Shrubsole <eshr@webrtc.org> Cr-Commit-Position: refs/heads/main@{#43788}
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.