7 Commits

Author SHA1 Message Date
Florent Castelli
8037fc6ffa Migrate absl::optional to std::optional
Bug: webrtc:342905193
No-Try: True
Change-Id: Icc968be43b8830038ea9a1f5f604307220457807
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/361021
Auto-Submit: Florent Castelli <orphis@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Florent Castelli <orphis@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#42911}
2024-09-02 12:16:47 +00:00
Markus Handell
97df932ecc Remove multiplex codec.
The feature isn't in use by Google and has proven to contain security
issues. It's time to remove it.

Bug: b/324864439
Change-Id: I80344eb2f2060469d2d69a54dc4519fdd02ab4ea
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/340324
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Commit-Queue: Markus Handell <handellm@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#41808}
2024-02-26 11:26:04 +00:00
qwu16
ae82df718c Add codec name H265 to support H265 in WebRTC
Bug: webrtc:13485
Change-Id: I352b15a65867f0d56fc8e9a9e03081bd3258108e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/316283
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#40773}
2023-09-20 09:25:32 +00:00
yingyingma
d3416972dd Video: add new metric for VP9/AV1 hw encoding with softwareBRC
Record the RMSE of encoding bitrate diff from output stream to target
bitrate set by WebRTC when the AV1/VP9 stream is removed.

Design doc: https://docs.google.com/document/d/1tE7Fquj1U6CwZ01QxxjoCaJS2o0LFYyIxvstLDx0qB0/edit#heading=h.c11ebhaja9tg

Bug: chromium:1358750
Change-Id: Ifa6f5aacfa8f6b0835f4609f102ae3ae1ee2897e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/298684
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39907}
2023-04-20 12:54:06 +00:00
Artem Titov
ab30d72b72 Use backticks not vertical bars to denote variables in comments for /video
Bug: webrtc:12338
Change-Id: I47958800407482894ff6f17c1887dce907fdf35a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/227030
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#34585}
2021-07-28 13:22:27 +00:00
Erik Språng
6c072efe9f Track both "network" and "media" rate in EncoderOvershootDetector
The network rate is based on a strict leaky buck with no underrun.
The media rate has almost the same algorithm, but allows the bufer level
to become negative down to -5 frames worth of data. This allows tracking
the average bitrate over a window and can be used as an alternative
upper bound when the encoder target is not link constrained.

Bug: webrtc:10155
Change-Id: I0d72157f7c0f0799e15572bf4797fa1b5636b22e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/129723
Commit-Queue: Erik Språng <sprang@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27392}
2019-04-01 13:10:49 +00:00
Erik Språng
7ca375c8ca Implement encoder overshoot detector and rate adjuster.
The overshoot detector uses a simple pacer model to determine an
estimate of how much the encoder is overusing the target bitrate.
This utilization factor can then be adjuster for when configuring the
actual target bitrate.

Spatial layers (simulcast streams) are adjusted separately.
Temporal layers are measured separately, but are combined into a single
utilization factor per spatial layer.

Bug: webrtc:10155
Change-Id: I8ea58dc6c4871e880553d7c22202f11cb2feb216
Reviewed-on: https://webrtc-review.googlesource.com/c/114886
Commit-Queue: Erik Språng <sprang@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26573}
2019-02-06 15:54:11 +00:00