Jakob Ivarsson db208317eb 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}
2023-01-27 15:46:32 +00:00
..
2022-10-08 08:38:36 +00:00
2022-09-09 07:33:11 +00:00