Commit graph

2 commits

Author SHA1 Message Date
Danil Chapovalov
6634c91194 Refactor AbsoluteCaptureTimeSender and AbsoluteCaptureTimeInterpolator
Removed thread safety: for a low level helper it adds overhead that users may not need. In particular RtpSenderVideo doesn't need it because calls to SendVideo are already synchronized.

Added a feature to force producing extension as requested by downstream.

Cleanup and document api:
Changed rtp_frequency type to int as it has no reason to use uint32_t per style guide
Changed absolute_capture_time to NtpTime to clarify both units and offset of the time. NtpTime has trivial conversion to/from uint64_t
Documented all the parameters.

Cleanup tests.

Bug: b/307553606
Change-Id: I0922ca4d3c89f124eeb561742dca79ed5c2327fd
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/325022
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Chen Xing <chxg@google.com>
Cr-Commit-Position: refs/heads/main@{#41023}
2023-10-27 12:50:08 +00:00
Minyue Li
63b3095d2b Make local to capturer clock offset a separate entry in PacketInfo.
This also changes the meaning of |estimated_capture_clock_offset| in
|absolute_capture_time_| to become a remote to capturer clock offset.

Bug: chromium:1056230, webrtc:10739
Change-Id: Id658590e027bbe77ae0834ea224e1dc977a305f2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/219163
Commit-Queue: Minyue Li <minyue@webrtc.org>
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Alessio Bazzica <alessiob@webrtc.org>
Reviewed-by: Christoffer Rodbro <crodbro@webrtc.org>
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Reviewed-by: Chen Xing <chxg@google.com>
Cr-Commit-Position: refs/heads/master@{#34067}
2021-05-20 13:42:57 +00:00