Commit graph

6 commits

Author SHA1 Message Date
Olga Sharonova
2d0ba28e25 Audio stack traces
Bug: webrtc:0
Change-Id: I90ea6301f02c2ebe72711ddbeda0bf000a6873aa
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/276940
Auto-Submit: Olga Sharonova <olka@webrtc.org>
Commit-Queue: Henrik Andreassson <henrika@webrtc.org>
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38223}
2022-09-27 15:05:51 +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
Artem Titov
913cfa76ec Use backticks not vertical bars to denote variables in comments for /modules/rtp_rtcp
Bug: webrtc:12338
Change-Id: I52eb3b6675c4705e22f51b70799ed6139a3b46bc
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/227164
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34686}
2021-08-09 15:51:03 +00:00
Markus Handell
e7c015e112 Migrate modules/rtp_rtcp to webrtc::Mutex.
Bug: webrtc:11567
Change-Id: I4c71f3a28ef875af2c232b1b553840d6e21649d3
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/178804
Commit-Queue: Markus Handell <handellm@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31645}
2020-07-07 12:13:47 +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
Chen Xing
9c16af7eb7 Add a tracker for RTCRtpContributingSource and RTCRtpSynchronizationSource.
This change adds a new SourceTracker class that can do spec-compliant tracking of RTCRtpContributingSource and RTCRtpSynchronizationSource when frames are delivered to the RTCRtpReceiver's MediaStreamTrack for playout. It will replace the existing spec-incompliant ContributingSources.

Bug: webrtc:10545 webrtc:10668
Change-Id: I961adaba09d6337f2f36b301a4fabcd20de65271
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/140948
Commit-Queue: Chen Xing <chxg@google.com>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28249}
2019-06-12 12:11:55 +00:00