commit | 75bbd1fbe6f9c19115bf847d7fd3977eabc1e558 | [log] [tgz] |
---|---|---|
author | Björn Terelius <terelius@google.com> | Wed Aug 25 11:46:07 2021 |
committer | WebRTC LUCI CQ <webrtc-scoped@luci-project-accounts.iam.gserviceaccount.com> | Wed Aug 25 11:46:34 2021 |
tree | bd43522c9c5081f3a9a63188ee790b1f257cb003 | |
parent | b7aac6f5f40730d0feb1d8a79d234daed4485cb8 [diff] |
Revert "red: generate and parse the red fmtp format" This reverts commit 9d0730942677a520ce7e184d081b4c5a2469fc48. Reason for revert: Speculative revert due to failing downstream test. If the test recovers, I'll assign the issue to the tests owners. Original change's description: > red: generate and parse the red fmtp format > > generates a fmtp line like > a=fmtp:<red payloadtype> <opus payloadtype>/<opus payloadtype> > and matches the incoming redundant payload types against the > send codec one. Offers without an FMTP line will not use RED. > Redundancy levels of 1 (plus main packet ) to 32 are accepted but > this is not wired up to the encoder since the O/A semantic of > RFC 2198 is not clear. > > This decreases the chance of a collision with the SATIN codec > which also runs on 48khz (but so far does not specify a channelCount of 2) > > BUG=webrtc:11640 > > Change-Id: I8755e5b1e944d105212f1bbe4f330cf4e0753e67 > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/229583 > Reviewed-by: Harald Alvestrand <hta@webrtc.org> > Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org> > Commit-Queue: Harald Alvestrand <hta@webrtc.org> > Cr-Commit-Position: refs/heads/main@{#34848} TBR=henrik.lundin@webrtc.org,hta@webrtc.org,philipp.hancke@googlemail.com,webrtc-scoped@luci-project-accounts.iam.gserviceaccount.com Change-Id: I5a0816a22a2a213679ab047c61e3b1dda40c4f59 No-Presubmit: true No-Tree-Checks: true No-Try: true Bug: webrtc:11640 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/230140 Reviewed-by: Björn Terelius <terelius@google.com> Commit-Queue: Björn Terelius <terelius@google.com> Cr-Commit-Position: refs/heads/main@{#34850}
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.