Commit graph

5 commits

Author SHA1 Message Date
Danil Chapovalov
f53597140f In RtpSource represent time with Timestamp type instead of int64_t
Bug: webrtc:13757
Change-Id: I5d7da9c9aee489e4b57d361de174c59713cb2b14
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/317780
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40650}
2023-08-29 10:05:03 +00:00
Alessio Bazzica
56b96ffe6a Surface local_capture_clock_offset from RtpSource
- Propagating `RtpPacketInfo::local_capture_clock_offset`, an
  existing field that is related to the abs-capture-timestamp
  header extension field `estimated_capture_clock_offset`
- Propagated through `SourceTracker::SourceEntry`

Bug: webrtc:10739, b/246753278
Change-Id: I21d9841e4f3a35da5f8d7b31582898309421d524
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/275241
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38129}
2022-09-20 12:51:22 +00:00
Alessio Bazzica
31996f48f4 RtpSource: remove deprecated ctor, use designated initializers
Bug: webrtc:10739, b/246753278
Change-Id: I215483709e1f415170bc42ea6d523ffad8eb1e76
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/275561
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38085}
2022-09-15 07:42:27 +00:00
Ruslan Burakov
d51cc7bd71 Add absolute capture time property to rtp sources.
This part of the effort to implement A/V sync metric.

Bug: webrtc:10739
Change-Id: I4adba1b99b37b31868168e37d9aa8e03f8ea6d4e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/159886
Commit-Queue: Ruslan Burakov <kuddai@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Reviewed-by: Minyue Li <minyue@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Ruslan Burakov <kuddai@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29849}
2019-11-20 18:50:45 +00:00
Niels Möller
a837030f8f Split out RtpSource from libjingle_peerconnection_api
And moved declaration into a new api directory, as
api/transport/rtp/rtp_source.h.

Bug: webrtc:8733
Change-Id: Ia73b7b0630e6065de4707a37633adddfa00a2b8a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/150880
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29039}
2019-09-02 14:04:47 +00:00