This is a reland of commit 7ae48c452a with updated RtpVp9RefFinder
RtpVp9RefFinder relied on the fact that frames with (inter_pic_predicted=true && inter_layer_predicted=true) were marked as keyframes. Since this is not the case anymore, the related code paths in RtpVp9RefFinder have been deleted.
Calculation of gof_info_[] index for non-keyframes has been updated to account for that fact it is now possible to received multiple T0 frames belonging to the same temporal unit (we don't need to do "unwrapped_tl0 - 1" in this case).
Original change's description:
> Mark frames with inter_layer_predicted=true as delta frames
>
> As it is currently implemented, the VP9 depacketizer decides packet's frame type based on p_bit ("Inter-picture predicted layer frame"). p_bit is set to 0 for upper spatial layer frames of keyframe since they do not have temporal refs. This results in marking packets of upper spatial layer frames, and, eventually these frames, of SVC keyframes as "keyframe" while they are in fact delta frames.
>
> Normally spatial layer frames are merged into a superframe and the superframe is passed to decoder. But passing individual layers to a single decoder instance is a valid scenario too and is used in downstream projects. In this case, an upper layer frame marked as keyframe may cause decoder reset [2] and break decoding.
>
> This CL changes frame type decision logic in the VP9 depacketizer such that only packets with both P and D (inter-layer predicted) bits unset are considered as keyframe packets.
>
> When spatial layer frames are merged into a superframe in CombineAndDeleteFrames [1], frame type of the superframe is inferred from the lowest spatial layer frame.
>
> [1] https://source.chromium.org/chromium/chromium/src/+/main:third_party/webrtc/modules/video_coding/frame_helpers.cc;l=53
>
> [2] https://source.corp.google.com/piper///depot/google3/third_party/webrtc/files/stable/webrtc/modules/video_coding/codecs/vp9/libvpx_vp9_decoder.cc;l=209
>
> Bug: webrtc:15827
> Change-Id: Idc3445636f0eae0192dac998876fedec48628560
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/343342
> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
> Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#41939}
Bug: webrtc:15827
Change-Id: Ic69b94989919cf6d353bceea85d0eba63bc500ee
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/344144
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41985}
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.
Development
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.
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