commit | 0145db40912babe3f74e00bed01df1a7b4502105 | [log] [tgz] |
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author | Henrik Boström <hbos@webrtc.org> | Wed Jul 26 12:01:59 2023 |
committer | WebRTC LUCI CQ <webrtc-scoped@luci-project-accounts.iam.gserviceaccount.com> | Wed Jul 26 13:48:41 2023 |
tree | b967b3ddce346b74d7f711d9b59b2d86f3a59f6c | |
parent | 9130431b54fd18aa4857614c886aeeed1731fcf7 [diff] |
Recreate the stream when switching from standard to legacy API. ReconfigureEncoder() is supposed to recreate the send stream when switching between legacy and standard API paths to ensure that the upper and lower layers agree on the number of streams that exist (legacy = 3 encodings but 1 stream, standard = same as encodings). This successfully happened when going from standard to legacy but due to a bug in the condition this did not happen when going from legacy to standard because `scalability_mode_used` is always false here (even though the standard path does use a scalability mode). As a consequence, SetRtpParameters()'s call to UpdateSendState() resulted in a DCHECK-crash. In release builds we still avoid IOOB because active_modules.size() < rtp_streams.size() but to avoid mistakes like this happening again in the future, the DCHECK is promoted to a CHECK. The fix is to remove the scalability mode condition which didn't make sense anyway - changing scalability mode does not require recreation but recreation is necessary when number of streams change, whether or not scalability mode changed. TESTED = Using Simulcast Playground and switching back and forth between standard and legacy and changing scalability modes and confirming from stats, see https://crbug.com/1467455. Bug: chromium:1467455 Change-Id: Ide29742972ba83f2e0a11f135ab9b39c39d4eb49 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/313280 Reviewed-by: Erik Språng <sprang@webrtc.org> Commit-Queue: Henrik Boström <hbos@webrtc.org> Cr-Commit-Position: refs/heads/main@{#40477}
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.