commit | 874517b5f57a68b17d4dabb449190bc5aab6fdae | [log] [tgz] |
---|---|---|
author | Henrik Boström <hbos@webrtc.org> | Tue May 31 12:22:41 2022 |
committer | WebRTC LUCI CQ <webrtc-scoped@luci-project-accounts.iam.gserviceaccount.com> | Tue May 31 13:31:56 2022 |
tree | bf8ed3ee52a4df50a9852febc38463a83a7f6959 | |
parent | 1a5a81340d81a783db748bc914af8d95047e0ab6 [diff] |
Revert "sdp: reject duplicate codecs with the same id but different name or clockrate" This reverts commit ad6807805d12e48f11c3a68b4befaf8d7c23e8b5. Reason for revert: Speculative revert due to consistent Mac browser test failures preventing WebRTC from rolling int Chromium: https://ci.chromium.org/ui/p/chromium/builders/webrtc.fyi/WebRTC%20Chromium%20FYI%20Mac%20Tester/10410/overview "Failed to parse SessionDescription. a=rtpmap:103 ISAC/16000 Duplicate payload type with conflicting codec name, clock rate or number of channels." Original change's description: > sdp: reject duplicate codecs with the same id but different name or clockrate > > since something like > rtpmap:96 VP8/90000 > rtpmap:96 VP9/90000 > or > rtpmap:97 ISAC/32000 > rtpmap:97 ISAC/16000 > is wrong. Note that fmtp or rtcp-fb are not taken into account. > Also note that sending invalid static payload types now throws an error. > > Drive-by: replace "RtpMap" with "Rtpmap" for consistency. > > BUG=None > > Change-Id: I2574b82a6f1a0afe3edc866e514a5dbca0798e8c > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/263641 > Reviewed-by: Harald Alvestrand <hta@webrtc.org> > Commit-Queue: Philipp Hancke <philipp.hancke@googlemail.com> > Cr-Commit-Position: refs/heads/main@{#37028} Bug: None Change-Id: Ic9c06c9309bb68bd94bfdd2e30ffd6ff96f6812b Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/264540 Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com> Commit-Queue: Henrik Boström <hbos@webrtc.org> Reviewed-by: Christoffer Jansson <jansson@webrtc.org> Reviewed-by: Philipp Hancke <philipp.hancke@googlemail.com> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Cr-Commit-Position: refs/heads/main@{#37064}
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.