This CL migrates unit tests to the new TaskQueueBase interface.
Bug: chromium:1416199
Change-Id: Ic15c694b28eb67450ac99fdd56754de1246a4d95
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/295621
Commit-Queue: Markus Handell <handellm@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39434}
This is a reland of commit 8ad4924936
See diff between latest Patch Set and PS1. Fixes include:
- VideoStreamEncoder's call to bitrate_adjuster_->OnEncodedFrame()
is updated to take stream index (spatial or simulcast index) instead
of only looking at SpatialIndex().
- Migrate test-only helpers to use Spatial/SimulcastIndex correctly.
The fixes are to migrate
some test-only helpers that we had forgot to fix that are used by
external tests.
Original change's description:
> Make SimulcastIndex() and SpatialIndex() distinct (remove fallback).
>
> This CL removes the fallback logic to return the other index when the
> one requested has not been set. This means we can remove the codec gates
> that was previously needed because SpatialIndex() had multiple meanings,
> resolving the TODOs previously added in
> https://webrtc-review.googlesource.com/c/src/+/293343.
>
> We have already migrated all known external dependencies from
> SpatialIndex() to SimulcastIndex() where necessary, unblocking this CL.
>
> PSA: https://groups.google.com/g/discuss-webrtc/c/SDAVg6xJ3gY
>
> Bug: webrtc:14884
> Change-Id: I82787505ab10be151e5f64965b270c45465d63a9
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/293740
> Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
> Reviewed-by: Erik Språng <sprang@webrtc.org>
> Commit-Queue: Henrik Boström <hbos@webrtc.org>
> Reviewed-by: Philip Eliasson <philipel@webrtc.org>
> Reviewed-by: Evan Shrubsole <eshr@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#39343}
Bug: webrtc:14884
Change-Id: Ib966924efca1a040dae881599f0789a7f2ab24a2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/294284
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Evan Shrubsole <eshr@webrtc.org>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39358}
This makes the max bitrate configured in RTP parameter to still be
respected, instead of being overridden by defaults in the case that an
encoder with untrusted QP is reconfigured (e.g. due to some adaptation).
Bug: webrtc:14914
Change-Id: I2d3ff645c069b80ec2e36887e6ce0ecd09a7ecbf
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/293944
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39357}
This reverts commit 8ad4924936.
Reason for revert: Breaks downstream projects
Original change's description:
> Make SimulcastIndex() and SpatialIndex() distinct (remove fallback).
>
> This CL removes the fallback logic to return the other index when the
> one requested has not been set. This means we can remove the codec gates
> that was previously needed because SpatialIndex() had multiple meanings,
> resolving the TODOs previously added in
> https://webrtc-review.googlesource.com/c/src/+/293343.
>
> We have already migrated all known external dependencies from
> SpatialIndex() to SimulcastIndex() where necessary, unblocking this CL.
>
> PSA: https://groups.google.com/g/discuss-webrtc/c/SDAVg6xJ3gY
>
> Bug: webrtc:14884
> Change-Id: I82787505ab10be151e5f64965b270c45465d63a9
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/293740
> Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
> Reviewed-by: Erik Språng <sprang@webrtc.org>
> Commit-Queue: Henrik Boström <hbos@webrtc.org>
> Reviewed-by: Philip Eliasson <philipel@webrtc.org>
> Reviewed-by: Evan Shrubsole <eshr@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#39343}
Bug: webrtc:14884
Change-Id: Ibcb834a1519930336fa50e8e9d8d0137972e28e6
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/294282
Owners-Override: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Auto-Submit: Henrik Boström <hbos@webrtc.org>
Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39347}
This CL removes the fallback logic to return the other index when the
one requested has not been set. This means we can remove the codec gates
that was previously needed because SpatialIndex() had multiple meanings,
resolving the TODOs previously added in
https://webrtc-review.googlesource.com/c/src/+/293343.
We have already migrated all known external dependencies from
SpatialIndex() to SimulcastIndex() where necessary, unblocking this CL.
PSA: https://groups.google.com/g/discuss-webrtc/c/SDAVg6xJ3gY
Bug: webrtc:14884
Change-Id: I82787505ab10be151e5f64965b270c45465d63a9
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/293740
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Reviewed-by: Evan Shrubsole <eshr@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39343}
When disabling a spatial layer, reconfiguration of the encoder is not
necessary (bitrate will never be assigned to the inactive layer anway).
This CL however makes sure we reconfigure the encoder when a spatial
layer is activated. Some encoder implementations may encoder the wrong
number of spatial layers if the active layers have not beens set
correctly.
Bug: webrtc:14809, b/261097903
Change-Id: I8d34aaec95eb50a9717c06ea38f25088e5a96429
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/290560
Commit-Queue: Philip Eliasson <philipel@webrtc.org>
Auto-Submit: Erik Språng <sprang@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38999}
At the same time, proper names of some parameters are refactored in SimulcastEncoderAdapter.
Bug: None
Change-Id: Ia036e3f362d1394e90aa26b79953c1ffe75e2fe0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/284961
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Commit-Queue: Chunbo Hua <chunbo.hua@intel.com>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38870}
As the synchronous version only posts a task to recreate the encoder
later, it is not possible to catch errors and state changes that
could appear then.
The asynchronous version of SetParameters() aims to solve this by
providing a callback to wait for the completion of the encoder
reconfiguration, allowing any error to be propagate and subsequent
getParameters() call to have up to date information.
Bug: webrtc:11607
Change-Id: I5548e75aa14a97f8d9c0c94df1e72e9cd40887b2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/278420
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Florent Castelli <orphis@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38627}
This cl/ implements configuring of encode resolution
in the video_stream_encoder (webrtc_video_engine) in
a way that is independent of frame resolution (i.e
not using scale_resolution_down_by).
The cl/ reuses the VideoAdapter as is, and hence
the output resolution will be the same as it is today.
Anticipated further patches
3) Hook up resource adaptation
4) Let VideoSource do adaption if possible
Bug: webrtc:14451
Change-Id: I881b031c5b23be26cacfe138730154f1cb1b66a8
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/276742
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Commit-Queue: Jonas Oreland <jonaso@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38245}
This cl/ is a NOP refactoring,
moving the EncoderStreamFactory from within webrtc_video_engine.cc
into own file in video/. simulcast.cc is collateral.
Bug: webrtc:14451
Change-Id: Ia69b9241d8cd8a12be6628d887701f2e244c07cc
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/276861
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Commit-Queue: Jonas Oreland <jonaso@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38224}
The Chromium RTCVideoEncoder unfortunately doesn't set if the
result is at target quality, and the definition of the threshold
is buried in libvpx_vp8_encoder.h.
This change
* Updates VideoStreamEncoder to postprocess an incoming EncodedImage
by interpreting the incoming QP information instead.
* Updates the related VideoStreamEncoder test to simulate an encoder
producing images around the QP threshold.
* Updates the steady state VP8 screencast QP threshold to a central
include file.
* Moves this and previously existing EncodedImage post-processing to a
new method AugmentEncodedImage.
Bug: b/245029833
Change-Id: I69ae29ffe501e84f28908f7d9a8cfd066ba82b43
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/275380
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Markus Handell <handellm@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38091}
The frame cadence adapter would previously crash on encountering
unconfigured layer updates. Fix this to silently ignore such updates.
Fixed: webrtc:14417
Change-Id: I524aa196f6aed10ffb8cd4ddcb4b053c71dfabba
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/273325
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Markus Handell <handellm@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37980}
We only need to see which bitrates have been configured, no need to
wait for failed frame. This should also reduce test durations somewhat.
Bug: None
Change-Id: Ie081310f9f80e21039c78d8c80510769cb400c3a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/270747
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37711}
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}
If a "normal" software buffer frame is dropped during paused state, we
store it as a pending frame and try encoding it after the pause state is
lifted. However, native frames are dropped entirely since keeping e.g.
texture handles for long time periods can lead to side effects.
Work around this by requesting a refresh frame after unpausing if the
dropped frame flag is set.
Bug: webrtc:14276
Change-Id: I9edd1e99454e082bcfe29f3d9041026dd8a390d0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/270220
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37660}
This CL accomplishes three things:
1) It enables feeding frame drop indications into the
AdaptedVideoTrackSource for the benefit of downstream projects.
2) Under zero hertz source delivery, a discarded frame ending a
sequence of frames which happened to contain important information
can be seen as a capture freeze. Avoid this by starting requesting
refresh frames after a grace period.
3) It changes the duration until first refresh frame requests on new
streams to three frame periods.
Bug: chromium:1324120, chromium:1336952
Change-Id: I0214852f1a26540588f6c193dd88a65c34ec0d99
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/265871
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Markus Handell <handellm@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37238}
Code that is probably bogus and causes frequent down-adaptation on
dual core systems was identified. We wish to remove this code in a
safe way. This CL achieves this under kill switch
WebRTC-MacSpecialOveruseRulesRemovalKillSwitch.
Fixed: webrtc:14138
Change-Id: Idf53348c8e1dc032d8eea58f626f91456d72ecb4
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/264423
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Evan Shrubsole <eshr@webrtc.org>
Commit-Queue: Markus Handell <handellm@webrtc.org>
Auto-Submit: Markus Handell <handellm@webrtc.org>
Commit-Queue: Evan Shrubsole <eshr@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37043}
In case the encoder TQ has been stopped and doesn't accept more tasks,
we could end up in a hung state during Stop(). This is a hypothetical
situation, but can be simulated in a test and avoided.
Bug: webrtc:14063
Change-Id: I20f48b11b6266f6875ed5e69de3529212505e439
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/258125
Reviewed-by: Evan Shrubsole <eshr@webrtc.org>
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36964}
This CL sets speed 9 for all resolutions when two or less cores are
available, as a heuristic for a "slow" machine.
This gives a large speed bost at a relatively small quality loss.
A field-trial kill-switch is available to override this behavior.
Bug: webrtc:13888
Change-Id: I24278a45de000ad7984d0525c47d9eb6b9ab6b60
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/257421
Reviewed-by: Emil Lundmark <lndmrk@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36466}
convert almost all of video/ (and the collateral)
Bug: webrtc:10335
Change-Id: Ic94e05937f54d11ee8a635b6b66fd146962d9f11
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/254601
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Jonas Oreland <jonaso@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36192}
Some of the state that's managed in VideoStreamEncoder, is updated
and accessed on the encoder queue, but a method that's used for testing
only (GetAdaptationResources()), represents a race between PostTask
operations that update state and checking for said state on the worker
thread.
This CL removes Wait() operations related to adaptation resources from
the common path and puts one in the test path instead.
Bug: webrtc:13612
Change-Id: Ie3e018e815e24951bc0634ed70de17eaf336a508
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/249220
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35797}
Currently `CreateLibaomAv1Encoder` will either return an actual libaom AV1 encoder or a nullptr depening on whether the build flag `enable_libaom` was configured to true or not. This CL updates the `libaom_av1_encoder` build target to no longer depend on `enable_libaom` so that `CreateLibaomAv1Encoder` will always return an encoder instance.
Added `CreateLibaomAv1EncoderIfSupported` as a replacement to the old `CreateLibaomAv1Encoder`.
Bug: webrtc:13573
Change-Id: Ibdcd52c609acd79feefa2b86f19d1b4ca3e91d0a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/242360
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Peter Hanspers <peterhanspers@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Commit-Queue: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35763}
Currently if encoder initialization fails WebRTC doesn't send any video.
This CL adds functionality that changes encoder type in such case and
restores the video. If encoder selector is available we switch to
encoder it recommends. Otherwise, VP8 is used as the default fallback
encoder.
Bug: webrtc:13572
Change-Id: Ifcdf707a575711f5ff81f9451caf30140c9171dc
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/246960
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35761}
The frame cadence adapter ignores key frame processing that happens
before the point where zero-hertz mode is activated, which leads to
no refresh frame requests if the key frame request comes too early.
Fix this to register a pending refresh frame request that gets
serviced when zero-hertz mode is activated. The CL changes the
FrameCadenceAdapterInterface::ProcessKeyFrameRequest from returning
whether to request a refresh frame into the frame cadence adapter
actively doing so itself via a new Callback::RequestRefreshFrame
API.
go/rtc-0hz-present
Bug: chromium:1255737
Change-Id: I53c2dbf6468e883eb2a2e81498e7134b1b35c336
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/242963
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Markus Handell <handellm@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35598}
Careful analysis of logs related to the requesting of refresh
frames from the source revealed an uncomfortable truth:
zero-hertz mode activates first when the first frame has been
received in the VideoStreamEncoder, because the number of simulcast
layers can only be computed when frame dimensions are known. This
fact means that the currently implemented logic for requesting
refresh frames is noneffective.
Fix this by
1. Activating zero-hertz mode prior of knowing the final layer
count. This causes refresh frame requests to happen without any
frames received from the source.
2. Making layer count dynamically configurable. Zero-hertz mode
considers layer quality unconverged after such a reconfiguration.
go/rtc-0hz-present
Bug: chromium:1255737
Change-Id: I0ecea4d2a8442a00e3b79b146dd39a5f4ac561d9
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/242860
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Markus Handell <handellm@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35593}
Under zero-hertz mode, provided that a frame arrived to the
VideoStreamEncoder, the receiver may experience up to a second
between incoming frames. This results in key frame requests getting
serviced with that delay, which is undesired.
What's worse is also the fact that if no frame ever arrived to the
VideoStreamEncoder, it will not service the keyframe requests at all
until the first frame comes.
This change introduces VideoSourceInterface::RequestRefreshFrame
which results in a refresh frame being sent from complying sources.
The method is used under zero-hertz mode from the VideoStreamEncoder
when frames didn't arrive to it yet (with changes to the zero-hertz
adapter).
With this change, when the frame adapter has received at least one
frame, it will conditionally repeat the last frame in response to the
key frame request.
go/rtc-0hz-present
Bug: chromium:1255737
Change-Id: I6f97813b3a938747357d45e5dda54f759129b44d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/242361
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Markus Handell <handellm@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35562}
The frame cadence adapter previously resulted in unconditional
frame repeating at max FPS. Change this to slow down to an idle
rate (1 Hz) when quality convergence in all configured spatial
layers has been achieved.
go/rtc-0hz-present
Bug: chromium:1255737
Change-Id: Ifa593dbf8a61aa29da20ac250da332734ae82791
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/241421
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Markus Handell <handellm@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35547}
The encoders wrapped in VideoStreamEncoder grossly over-estimates
available bitrate when capture FPS falls close to zero, and frames
re-commence highly frequent delivery. Avoid this by moving the input
RateStatistics inside VSE into the frame cadence adapter, and changing
the reported framerate under zero-hertz encoding mode to always return
the configured max FPS.
Bug: chromium:1255737
Change-Id: Iaa71ef51c0755b12e24e435d86d9562122ed494e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/239126
Commit-Queue: Markus Handell <handellm@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35431}
This change switches the sequence used by the FrameCadenceAdapter
to be the encoder_queue, enabling VideoStreamEncoder::OnFrame to be
invoked directly on the encoder_queue and eliminates the contained
PostTasks.
Bug: chromium:1255737
Change-Id: Ib86fc96ad2be9a38585fef2535855e3f9cc7e57c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/238171
Commit-Queue: Markus Handell <handellm@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35380}
Add implementation of RTC_DCHECK_NOTREACHED equal to the RTC_NOTREACHED.
The new macros will replace the old one when old one's usage will be
removed. The idea of the renaming to provide a clear signal that this
is debug build only macros and will be stripped in the production build.
Bug: webrtc:9065
Change-Id: I4c35d8b03e74a4b3fd1ae75dba2f9c05643101db
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/237802
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35348}
VideoStreamEncoder receives frames on an undefined threading
context with the only requirement being that frames are serially
arriving. This CL changes this to post all frames arriving at the
FrameCadenceAdapter to the worker thread before further
processing, transitively leading to frame entry into the
VideoStreamEncoder on the worker thread.
Bug: chromium:1255737
Change-Id: I04d69cb4a5048d671d2dcd3bd6d669fbcda52b3f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/237142
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Markus Handell <handellm@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35320}
This change introduces a new FrameCadenceAdapter class which takes the
role of being a VideoFrameSinkInterface<> instead of VideoStreamEncoder.
The FrameCadenceAdapter will see its functionality grow in future CLs
and eventually enable screenshare capture sources to have zero hertz as
the minimum capture frequency.
This CL moves logic related to UMA collection and constraints into the
adapter.
The adapter has two major modes. Future functionality is planned to be
added under the WebRTC-ZeroHertzScreenshare field trial. Unit tests are
added that verify passthrough operation when WebRTC-ZeroHertzScreenshare
isn't specified or disabled.
Just specifying the WebRTC-ZeroHertzScreenshare field trial isn't
enough to activate the feature, but the caller has to additionally
configure screen content type, minimum FPS 0, and maximum FPS > 0 for
the new mode.
go/rtc-0hz-present
Bug: chromium:1255737
Change-Id: I1799110ed40843152786ad80df10acfb83a608b1
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/236682
Commit-Queue: Markus Handell <handellm@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35315}
