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5 commits

Author SHA1 Message Date
Rasmus Brandt
f0820ffd88 Implement video versions of RTCInboundRtpStreamStats.jitterBuffer{Target,Minimum}Delay
* https://www.w3.org/TR/webrtc-stats/#dom-rtcinboundrtpstreamstats-jitterbuffertargetdelay
* https://www.w3.org/TR/webrtc-stats/#dom-rtcinboundrtpstreamstats-jitterbufferminimumdelay

Tested: https://jsfiddle.net/pfgzj0yo/17/

Bug: webrtc:14244
Change-Id: I3d949ba63c8339b3881f5d00356559d5789d283d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/304404
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40157}
2023-05-26 13:34:09 +00:00
Philipp Hancke
a204ad210d clean up misc TimeDelta use
follow-up from https://webrtc-review.googlesource.com/c/src/+/262810

* replace Time::Millis(0) and TimeDelta::Millis(0) with ::Zero()
* drop unnecessary webrtc namespace from some TimeDeltas
* make TimeDelta do the unit conversion for stats

BUG=webrtc:13756

Change-Id: Ic60625ae0fc7959a47a6be9f5051851feaf76373
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/265875
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37664}
2022-08-02 13:52:36 +00:00
Evan Shrubsole
496ad52cb2 UpdateCurrentDelay should update delay with rounding errors
Some of the timestamps input into UpdateCurrentDelay are not truncated
to milliseconds and thus a small negative delay can result. This means
the delay will not update when it should have.

Bug: webrtc:14168
Change-Id: I5293339b6a39201c680854e9596b717025ee8dc1
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/266370
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Evan Shrubsole <eshr@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37657}
2022-08-02 08:47:54 +00:00
Johannes Kron
bbf639e930 Add low-latency stream signaling to VideoFrame and VCMTiming
This is the first CL out of three to make the low-latency stream signaling
explicit. At the moment this is done by setting the render time to 0.
There's a dependency between Chromium and WebRTC which is why this is
split into three CLs to not break any existing functionality.

Bug: chromium:1327251
Change-Id: Ie6b268746d587a99334485db77181fb2c6e9b567
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/264502
Reviewed-by: Evan Shrubsole <eshr@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Johannes Kron <kron@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37225}
2022-06-15 14:04:28 +00:00
Rasmus Brandt
c4d253c1ed Move VCMTiming into timing sub-folder
Bug: webrtc:14111
Change-Id: I9785b00012ea84f55789845a7e71fe26006d5067
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/263581
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Commit-Queue: Rasmus Brandt <brandtr@webrtc.org>
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37061}
2022-05-31 13:01:24 +00:00
Renamed from modules/video_coding/timing_unittest.cc (Browse further)