tree fcd8fc01bb968a3a129baa0acd086f252cb8d356
parent 9d21eb3e3eb279d587f9726a7a03e168b688e340
author Jakob Ivarsson <jakobi@webrtc.org> 1674828802 +0100
committer WebRTC LUCI CQ <webrtc-scoped@luci-project-accounts.iam.gserviceaccount.com> 1674834392 +0000

Update RTP timestamp based on capture timestamp when audio send stream is resumed.

This removes the previous approach where we continued to update the timestamp when the capturer is running but the send stream is stopped in favor of a more general approach that also works when the capturer is paused.

Some assumptions for this change to be correct: input sample rate and frame size will be the same before/after the stream is paused.

Bug: webrtc:12397
Change-Id: I3b03741cd6d3285cbc9aee3893800729852e6cfa
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/291526
Commit-Queue: Jakob Ivarsson‎ <jakobi@webrtc.org>
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39213}