This commit fixes the issue of video playback in slow motion caused by VCMTiming being unable to provide the correct rendering time in
scenarios of continuous network packet loss
WANT_LGTM=mbonadei
Bug: webrtc:376183208
Change-Id: I63617068506e536c4b812215ea084eec18e8ee06
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/367000
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43392}
The CodecTimer is not a codec timer, it's more like a decoder stopwatch with a percentile filter wrapped around it. Since the purpose of the class is to provide an estimate for how much decode delay to add when determining the render timestamp of a frame, let's rename this class to `DecodeTimePercentileFilter`.
No functional changes are intended.
Bug: webrtc:14905
Change-Id: I48c99e4f500c4f9e1a2a20b0afe72d6e76c5192d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/293462
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Commit-Queue: Rasmus Brandt <brandtr@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39332}
The `TimestampExtrapolator` is only used by the `VCMTiming`
class, despite there being references to it from both
`modules/rtp_rtcp/BUILD.gn` and `modules/video_coding/BUILD.gn`.
Bug: webrtc:14111
Change-Id: If1a02a56a0c83b13d619ca08dc76c884fa829369
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/275482
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Johannes Kron <kron@webrtc.org>
Commit-Queue: Rasmus Brandt <brandtr@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38093}
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}
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}
