commit | 949d3c9acf59af6bdcdf5152cac2df9cd5925cc3 | [log] [tgz] |
---|---|---|
author | Philipp Hancke <phancke@meta.com> | Thu Sep 26 17:20:10 2024 |
committer | WebRTC LUCI CQ <webrtc-scoped@luci-project-accounts.iam.gserviceaccount.com> | Mon Sep 30 18:03:49 2024 |
tree | a5f1274bf0e6a70fc897bfe6f5754bc7e0484145 | |
parent | 4773539f9070bb5e430a5353c22f4dbc2b11590a [diff] |
Reland "h264: fix first_packet_in_frame logic for multislice in a single rtp packet" This reverts commit bdc669347c70160cd648f5cab7a417227d41d82a. Reason for revert: AUDs will be taken into account now. video_replay with the provided out.pcap and these options: --codec H264 --input_file out.pcap --media_payload_type 102 --ssrc 40000 plays smoothly. Original change's description: > Revert "h264: fix first_packet_in_frame logic for multislice in a single rtp packet" > > This reverts commit 3753c8190e3f0aca6758a5521e33f8b5d4f09ab4. > > Reason for revert: Break assembling of hardware encoded h264 P frame on > weak network condition. > > Original change's description: > > h264: fix first_packet_in_frame logic for multislice in a single rtp packet > > > > a frame must be (or should be) first when it contains either SPS (but not just PPS), > > is an IDR or is a slice with first_mb_in_slice == 0. > > > > Fixes an edge case where a STAP-A with SPS, PPS and multiple slices of an IDR fit > > into a single RTP packet which can happen with small 320x196 frames > > > > BUG=webrtc:352379280,webrtc:346608838 > > > > Change-Id: Ic6dea6c81db759d0d7ddd4054407103fd791f6c5 > > No-Try: true > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/357121 > > Reviewed-by: Danil Chapovalov <danilchap@webrtc.org> > > Commit-Queue: Sergey Silkin <ssilkin@webrtc.org> > > Reviewed-by: Sergey Silkin <ssilkin@webrtc.org> > > Cr-Commit-Position: refs/heads/main@{#42652} > > Bug: webrtc:368335257 > Change-Id: I07725c78be628bff71b79b8b9369677e39f5f5ac > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/363080 > Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> > Reviewed-by: Sergey Silkin <ssilkin@webrtc.org> > Reviewed-by: Danil Chapovalov <danilchap@webrtc.org> > Reviewed-by: Philipp Hancke <phancke@meta.com> > Cr-Commit-Position: refs/heads/main@{#43062} Bug: webrtc:368335257 Change-Id: Idfae2efc1ebd7b97a2f7ebbd9d1e8c7bf6fcc348 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/363842 Reviewed-by: Sergey Silkin <ssilkin@webrtc.org> Commit-Queue: Philipp Hancke <phancke@meta.com> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/main@{#43113}
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.