commit 72a612020d8b779e9deb44cd334f90de1761f629
author Rasmus Brandt <brandtr@webrtc.org> Fri Oct 03 13:42:46 2025
committer WebRTC LUCI CQ <webrtc-scoped@luci-project-accounts.iam.gserviceaccount.com> Fri Oct 03 14:55:16 2025
tree 77256c35962dd20c927c593f9aeb612f47482287
parent f938898734b36d72a81794cecf838693a4293cf5

Create `video/timing` dir and implement `RtpPacketSimulator`.

This CL is the first in a chain, where the end goal is to create a video timing simulator. The simulator will take a `ParsedRtcEventLog` and produce a sequence of "rendered" frames, where the timing of the rendering should be completely determined by the logged RTP packets and the video jitter buffer components.

Big picture can be seen here: https://webrtc-review.googlesource.com/c/src/+/413840

In this CL, I'm creating the folder structure and implementing the `RtpPacketSimulator`. The `RtpPacketSimulator` is responsible for creating an `RtpPacketReceived` from a `LoggedRtpPacketIncoming`.

I'm also creating a new `rtc_test` binary (called `video_timing_tests`) that will contain all the unit tests for the upcoming simulator. For that, I followed https://webrtc.googlesource.com/src/+/refs/heads/main/g3doc/add-new-test-binary.md.

Tested: https://ci.chromium.org/ui/p/webrtc/builders/try/linux_rel/86942/test-results?q=RtpPacketSimulator+
Bug: b/423646186
Change-Id: I199e296f7bf2374f6f07638af3a23575c2a116cc
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/413700
Reviewed-by: Jeremy Leconte <jleconte@google.com>
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Commit-Queue: Rasmus Brandt <brandtr@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#45822}