As mentioned in https://crbug.com/webrtc/11956, the results did not show
any performance improvments.
Bug: webrtc:11956
Change-Id: Ie050aa5a6083fcf0c776fb8d03e7d18644b37f97
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/272280
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Commit-Queue: Evan Shrubsole <eshr@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37833}
This was only set to nullptr in non-test environments and was thusly
unused. With this change, the stats callbacks are gaurenteed to only
come from the VideoStreamBufferController and so the thread checks can
be removed.
Bug: webrtc:14003
Change-Id: Iaf0e77aa7c45a317e38ae27739edcefd3123d832
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/272021
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Commit-Queue: Evan Shrubsole <eshr@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37816}
I've added the proper headers to the only file in Chromium which includes screen_capture_frame_queue.h (see https://chromium-review.googlesource.com/c/chromium/src/+/3836317).
I've also built the remoting host and Chrome on Windows and Linux with this change and did not see any build errors.
The only build error I encountered was in shared_screencast_stream when building webrtc so I added the required header there.
Bug: webrtc:14378
Change-Id: Ie88e606dfa52f18514a87b87e5904424543d7df3
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/271922
Commit-Queue: Joe Downing <joedow@google.com>
Reviewed-by: Alexander Cooper <alcooper@chromium.org>
Cr-Commit-Position: refs/heads/main@{#37811}
Moves FrameBuffer2 to its own GN target to reduce the binary size of the
video target.
Bug: None
Change-Id: I40e86a1eabc0c9e8e6fada3dcdb4e3a043c61c6c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/271286
Commit-Queue: Evan Shrubsole <eshr@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37803}
- Move existing check on `max_frame_size` to the top.
By doing this early, the filter will not end up in an
inconsistent state (predicted but not updated) when
called with a tiny `max_frame_size`.
- Add sanity check on noise variance.
This will avoid sqrt of a negative number.
Bug: webrtc:14151
Change-Id: I2507a9114655d3c3ae35bf5ed83f3f1154c42ad3
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/271281
Commit-Queue: Rasmus Brandt <brandtr@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37798}
In migrating rtc::Event to use TimeDelta instead of int,
rtc::Event::kForever will have to become something else.
This change removes dependencies on that kForever is int.
Bug: webrtc:14366
Change-Id: Ic36057dda95513349e7ae60204e7271ff1f58825
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/271288
Auto-Submit: Markus Handell <handellm@webrtc.org>
Reviewed-by: Magnus Flodman <mflodman@webrtc.org>
Commit-Queue: Magnus Flodman <mflodman@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37795}
This CL reorders the update steps physically, to make them
easier to comprehend. It renames variables to be more verbose,
but also adds succinct mathematical descriptions (using Wikipedia
notation) to all steps.
No functional changes are intended with this change.
Bug: webrtc:14151
Change-Id: I6a4642e89e2b73639f0b4c928e07b317c14d5884
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/271546
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Commit-Queue: Rasmus Brandt <brandtr@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37784}
This class is never overridden.
Bug: webrtc:14151
Change-Id: I3b70e927ee0eafd71ce4bdb8f6e8d6330c1a3f08
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/271501
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Commit-Queue: Rasmus Brandt <brandtr@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37781}
* Update naming of data members.
* Start reordering code blocks in `PredictAndUpdate`.
(The "predict" step is done in this C:. The "update"
step will be improved in another cl.)
Bug: webrtc:14151
Change-Id: Idea1e8e786bd672dadedbcb3cd5598f4a033e81e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/271023
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Commit-Queue: Rasmus Brandt <brandtr@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37767}
This CL simplifies and documents the interface of the Kalman filter better. A simple unit test verifying the filter's convergence is
added. No functional changes to the filter are intended.
Future CLs will improve the data member naming and code organization.
Bug: webrtc:14151
Change-Id: I01e60d4b783cabad3ccbdc711c5d746666dd16ca
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/265877
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Commit-Queue: Rasmus Brandt <brandtr@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37766}
Instead of showing individual byte differences, this CL detects
differences in the expected and actual byte streams of the evaluated
AEC dump and, if detected, parses the `audioproc::Event` proto lite
messages and calls EXPECT_EQ() for a subset of individual (sub-)fields.
Note that messages are parsed only if the byte streams of each message
pair do not match, so with no failures the test runs at no extra cost.
Plus, the the added funcionality can only be enabled for local
debugging by flipping the `kDumpWhenExpectMessageEqFails` flag - a
code change cannot land if the flag is set to true.
Note that `MessageDifferencer` (see [1]) could not be used because
it is not implemented for `MessageLite` protos.
[1] https://developers.google.com/protocol-buffers/docs/reference/cpp/google.protobuf.util.message_differencer
Bug: b/241923537
Change-Id: I8e0eda3b1ecfe06945b6dad5ee8871f8200d76d7
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/270922
Reviewed-by: Jesus de Vicente Pena <devicentepena@webrtc.org>
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37765}
When ScreencastPortal::OnStartRequestResponseSignal receives either a
non-zero response code or is missing the response data, it would
directly cast this to a RequestResponse. However, this direct cast is an
error. Per the documentation, the response signal returns the following
values with their corresponding meanings:
0 - Success
1 - User Cancelled
2 - Error
The RequestResponse enum however, has "kUnknown" as 0, and thus
"kSuccess" as 1 (with all other values also shifted up by 1 value). This
means that when the portal was cancelled, we were still receiving
RequestResponse::kSuccess. This fixes the issue by removing the improper
cast and adding a translation function. This function is local for now
since no where else attempted to cast values to a RequestResponse; but
can be moved if the need arises.
Fixed: chromium:1351824
Change-Id: I4cd44d90055147c9592d590c7969dcfc3297a3d9
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/271240
Commit-Queue: Alexander Cooper <alcooper@chromium.org>
Auto-Submit: Alexander Cooper <alcooper@chromium.org>
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37755}
that rtc::Location parameter was used only as extra information for the
RTC_CHECKs directly in the function, thus call stack of the crash should
provide all the information about the caller.
Bug: webrtc:11318
Change-Id: Iec6dd2c5de547f3e1601647a614be7ce57a55734
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/270920
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37748}
Remove check if `prev_estimate_` is less than 10 us since it can never
be less than 1 ms.
Bug: None
Change-Id: If151390d22fa0b6ecdc36af64168d3e2049c7b6b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/271203
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Commit-Queue: Johannes Kron <kron@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37745}
The data that's used to report the histograms is owned by UlpfecReceiver
and moving the reporting there, simplifies things as configuration
changes happen in RtpVideoStreamReceiver2 (which currently require all
receive streams to be deleted+reconstructed).
Additional updates:
* Consistently using `Clock` for timestamps. Before there was
a mix of Clock and rtc::TimeMillis.
* Update code to use Timestamp and TimeDelta.
Bug: none
Change-Id: I89ca28ec7067a49d6b357315ae733b04e7c5a2e3
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/271027
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37729}
replace std::deque implementation with a manually controlled circular buffer.
replace Timestamp validity check from 'IsInfinite()' accesser to cheaper comparison to zero.
These greatly increase PacketArrivalTimeMap::AddPacket perfomance when packet arrive with large sequence number gaps.
Bug: chromium:1349880
Change-Id: I6f4e814b1086ca9d0b48608531e3a387d9e542dc
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/270564
Reviewed-by: Victor Boivie <boivie@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37722}
The intention of this change is to separate the Kalman filter state
(that prior to this change lived in JitterEstimator) from the
other filter's state, making it easier to see how the different
filters interact.
This move does not include any interface, functional, or
documentation changes. Those will follow in later changes.
A very basic unit test is added, which will also be expanded
later on.
Bug: webrtc:14151
Change-Id: Ifb9b8ce2d9418ea52ccf64a77fd46d1ebba30779
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/264984
Commit-Queue: Rasmus Brandt <brandtr@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37721}
We pass the fd we recieve from xdg-desktop-portal to PipeWire to connect
to it and according to the specification PipeWire automatically closes
it on disconnect or failure. We also close the fd ourself when we tear
down the portal connection so we have to avoid doing this twice. Looks
OBS studio just duplicates the fd passed to PipeWire so do the same in
order to avoid the fd ownership violation once we stop sharing.
The fd we recieve from xdg-desktop-portal is from PipeWire also using
fcntl() with F_DUPFD_CLOEXEC option.
Bug: chromium:1339236
Change-Id: Ia7aee36e520dd5ff9a40688a6807e31c4e636f8e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/270421
Commit-Queue: Alexander Cooper <alcooper@chromium.org>
Reviewed-by: Alexander Cooper <alcooper@chromium.org>
Cr-Commit-Position: refs/heads/main@{#37712}
* Add test to Generic decoder unittests to ensure drop behaviour is covered.
* Use simulated time in the generic decoder unittests.
Bug: webrtc:14324
Change-Id: I10b28b45c434f92d5344683fb9ca6676efe0e08c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/270662
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Evan Shrubsole <eshr@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37710}
Sample rates not divisible by 100, in particular 11025 Hz and 22050 Hz, have long been used with APM in Chrome, but the support has never been stated explicitly.
This CL makes minor modifications to the APM API to clarify how rates are handled when 10 ms is not an integer number of samples. Unit tests are also extended to cover this case better.
This does not update all references to 10 ms and implicit floor(sample_rate/100) computations, but it does at least take us closer to a correct API.
Note that not all code needs to support these sample rates. For example, audio processing submodules only need to operate on the native APM rates 16000, 32000, 48000.
Bug: chromium:1332484
Change-Id: I1dad15468f6ccb9c0d4d09c5819fe87f8388d5b8
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/268769
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Commit-Queue: Sam Zackrisson <saza@webrtc.org>
Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37682}
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}
This should avoid any potential race in static initialization for the
tracing logic.
Bug: webrtc:12715
Change-Id: Ic91d8e5fbd9b45a91e7e7a9e76226fc558e00c4b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/270381
Auto-Submit: Erik Språng <sprang@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37662}
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}
Before this change the full screen application handler was failing to
detect PowerPoint going into presentation mode, resulting in the editor
window continuing to be shared rather than the intended behavior of
sharing the presentation itself.
Fix this by always looking for the PowerPoint full screen presentation
window, regardless of whether the editor window is still open. In
the current version of PowerPoint, the editor stays open during
presentation.
Bug: chromium:1231437
Change-Id: I1b21e263d25320cc236d127d22d4d64bb52fcbda
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/269560
Reviewed-by: Mark Foltz <mfoltz@chromium.org>
Commit-Queue: Mark Foltz <mfoltz@chromium.org>
Cr-Commit-Position: refs/heads/main@{#37632}
This is the first step of migrating
AudioProcessing::CreateAndAttachAecDump() from using std::string to
absl::string_view.
Bug: webrtc:13579
Change-Id: I8fc373e7ac55fd8e96bb0b01d1a30e28883ac9a2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/269400
Commit-Queue: Ali Tofigh <alito@webrtc.org>
Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37631}
This overload was removed in C++20.
Bug: chromium:1284275
Change-Id: I67a25ae23fa111e4972d1b207f1c078da13d86a3
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/269440
Reviewed-by: Mark Foltz <mfoltz@chromium.org>
Auto-Submit: Peter Kasting <pkasting@chromium.org>
Commit-Queue: Peter Kasting <pkasting@chromium.org>
Cr-Commit-Position: refs/heads/main@{#37630}
WebRTC doesn't produces such packet and ignores it when receive.
Bug: None
Change-Id: I4af8cb3308cb2422808bdfc420a85fa175085bfb
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/269181
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37627}
instead of using Lock/Unlock attributes, use Assert attribute to annotate code is running on certain task queue or thread.
Such check better matches what is checked, in particular allows to
recheck (and thus better document) currently used task queue
Bug: None
Change-Id: I5bc1c397efbc8342cf7915093b578bb015c85651
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/269381
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37619}
Replace helper functions with the constant
Remove option to set min bitrate in RemoteBitrateEstimator as unused:
ReceivedSideCongestionController is the only user of the
RemoteBitrateEstimator interface, and it always sets the same value
right after construction that RemoteBitreateEstimators already use.
Bug: None
Change-Id: If179fdd72b1ded6ad1fd0a6dfffc97b302153322
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/269383
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37613}
Keeping the field trial around as a kill-switch for another milestone.
Bug: webrtc:11340
Change-Id: I3285baefab176f541cbb5ed3bacbc669d3e8836f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/269384
Commit-Queue: Erik Språng <sprang@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37611}
Inserting packet with zero arrival time may trigger inconsistent state in the internal map where packet sometimes treated as received, but sometimes treated as not received.
Bug: chromium:1346959
Change-Id: I0809e41a873103dcd62528358e64794c1d3cb28f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/269382
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37609}
1. Add loss threshold for high bandwidth preference. If the average loss ratio is less than the threshold, then the model prefers higher bandwidth candidates. Otherwise, it prefers lower bandwidth candidates. Before, it always prefers higher bandwidth candidate. The default value is 0.99, means it always prefers high bandwidth candidates.
2. Only increase the estimate if the inherent loss (random loss) is equal to/greater than the average loss. If the inherent loss is less than the average loss, then it is oversending, thus should not increase the estimate.
Bug: webrtc:12707
Change-Id: I37eb536679ca29e017a4a47703b417efd4d103ca
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/269101
Commit-Queue: Diep Bui <diepbp@webrtc.org>
Reviewed-by: Jakob Ivarsson <jakobi@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37608}
Consumers expect the frame to be valid if Result::SUCCESS is delivered.
If the frame is nullptr, we should deliver ERROR_TEMPORARY instead.
Bug: webrtc:14265
Change-Id: If94a3ead38d7657d7b90bbe046256be697312216
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/269223
Commit-Queue: Mark Foltz <mfoltz@chromium.org>
Reviewed-by: Mark Foltz <mfoltz@chromium.org>
Cr-Commit-Position: refs/heads/main@{#37590}
Update an affected unit test by the change in goog_cc.
Bug: webrtc:14272
Change-Id: I83e97530c861b126bed876d57f6d4f91aa45de7e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/269002
Reviewed-by: Jakob Ivarsson <jakobi@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Commit-Queue: Diep Bui <diepbp@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37587}
It is now easier to fully test `AgcManagerDirect` with different values
for the used field trials. In particular, this CL adds tests for the
field trial named `WebRTC-Audio-2ndAgcMinMicLevelExperiment`.
1. `UnmutingRaisesTooLowVolume` and `MicVolumeIsLimited`
The expectations for the lowest input volume are not hard-coded anymore
since the parametrized tests use different values for the enforced
minimum.
2. `RecoveryAfterManualLevelChangeBelowMin`
The recovery behavior after manual input volume change depends on
whether the minimum input volume is overridden. When that's the case,
the minimum volume is applied immediately after the manual adjustment.
Hence, the existing test is left and a parametrized version of it has been added to test the "instant recovery" behavior. The latter test is
skipped when the minimum input volume is not overridden since that case
is covered by the existing test.
Bug: chromium:1275566
Change-Id: Ib0d4427b32b88f33138d4062b365916a3c47a406
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/268900
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Reviewed-by: Hanna Silen <silen@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37577}
Using 4 temporal layers is not quite supported: Not advertised, no
integration tests. When transitioning to configuration via scalability
mode, there are no corresponding modes defined. So delete these two
tests; they can be added back if/when support for corresponding
scalability modes are added.
Bug: webrtc:11607
Change-Id: I97f55dc95d6513ccf65fa887757a62e9c8659be7
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/269003
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37575}
Stop using TEST_F; that will make it easier to switch to parametric
tests that are needed to correctly test `AgcManagerDirect`.
"Avoid fixtures where reasonable."
Source: https://abseil.io/tips/122
Bug: chromium:1275566
Change-Id: I2d73a0913eb2349144f63bd17ab4d6efa245e472
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/268766
Reviewed-by: Hanna Silen <silen@webrtc.org>
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37556}
rtc::TaskQueue is a simple wrapper over TaskQueueBase and adds no
extra features when task queue is used without passing ownership.
Reducing usage of the internal rtc::TaskQueue wrapper gives users more flexibility how TaskQueueBase* is stored.
Bug: webrtc:14169
Change-Id: If5c8827544c843502c7dfcef775ac558de79ec3a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/268189
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37549}
This reverts commit d0a6fd239c.
Reason for revert: reland the bug fix
Original change's description:
> Revert "`AgcManagerDirect`: stop enforcing min mic level override with 0 level"
>
> This reverts commit e76daab8b3.
>
> Reason for revert: revert required to revert the parent CL
>
> Original change's description:
> > `AgcManagerDirect`: stop enforcing min mic level override with 0 level
> >
> > https://webrtc-review.googlesource.com/c/src/+/250141 introduced a bug
> > due to which the min mic level override is always enforced, if specified
> > even if the user manually adjusts the mic level to zero.
> >
> > This CL fixes that bug, the changes run behind a kill switch.
> >
> > TESTED=Test video call on Chromium on Mac; input volume not adjusted after zeroing it from the system preferences UI
> >
> > Bug: chromium:1275566
> > Change-Id: I18ce2e5970d3002b301f51f84544583c64982d57
> > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/267844
> > Reviewed-by: Hanna Silen <silen@webrtc.org>
> > Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
> > Cr-Commit-Position: refs/heads/main@{#37460}
>
> Bug: chromium:1275566
> Change-Id: I6d22d8f3fafdc7da3814827b9b69146a506595db
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/268468
> Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
> Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#37515}
Bug: chromium:1275566
Change-Id: I7198587dec2a153270e8beb714e9dacccdaae806
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/268544
Reviewed-by: Hanna Silen <silen@webrtc.org>
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37530}
This reverts commit c9cad23274.
Reason for revert: add back field trial
Original change's description:
> Min mic analog level: override minimum and behavior on Mac
>
> This CL removes the `WebRTC-Audio-AgcMinMicLevelExperiment` field trial
> and always enables the code path behind that flag on Mac. In summary,
> the analog AGC behaves as follows on Mac:
> 1. the minimum level is overridden to 20
> 2. the minimum is applied even when clipping is detected
> 3. when the level is manually adjusted to 0, the minimum level is
> enforced - i.e., 20
>
> Note that the 3rd property had been unintentionally added when the
> changes were added behind the aforementioned field trial. This will
> be fixed in a follow-up CL.
>
> Bug: chromium:1275566
> Change-Id: If184c4455a0780fcd94f55141af34460c152e3c3
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/266488
> Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
> Reviewed-by: Hanna Silen <silen@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#37459}
Bug: chromium:1275566
Change-Id: I00a37ad9e16efc49f721558d25af16efd5f3db8c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/268540
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Reviewed-by: Hanna Silen <silen@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37521}
This reverts commit e76daab8b3.
Reason for revert: revert required to revert the parent CL
Original change's description:
> `AgcManagerDirect`: stop enforcing min mic level override with 0 level
>
> https://webrtc-review.googlesource.com/c/src/+/250141 introduced a bug
> due to which the min mic level override is always enforced, if specified
> even if the user manually adjusts the mic level to zero.
>
> This CL fixes that bug, the changes run behind a kill switch.
>
> TESTED=Test video call on Chromium on Mac; input volume not adjusted after zeroing it from the system preferences UI
>
> Bug: chromium:1275566
> Change-Id: I18ce2e5970d3002b301f51f84544583c64982d57
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/267844
> Reviewed-by: Hanna Silen <silen@webrtc.org>
> Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#37460}
Bug: chromium:1275566
Change-Id: I6d22d8f3fafdc7da3814827b9b69146a506595db
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/268468
Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37515}
There is a AV where GDI capturer sends empty frame to the
blank detector. It is fine operation from the GDI capturer
to pass an empty to the next handler. So, blank capturer
filter it and send it as blank frame to next handler.
Bug: webrtc:14265
Change-Id: Ifc90a210703e14fa6d0dc7fb2ae2942ae4e8125f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/268444
Reviewed-by: Alexander Cooper <alcooper@chromium.org>
Commit-Queue: Alexander Cooper <alcooper@chromium.org>
Cr-Commit-Position: refs/heads/main@{#37513}
This reverts commit 791294a647.
Reason for revert: downstream test adjusted
Original change's description:
> Revert "Fix overflow due to rounding in AbsoluteSendTime::To24Bits"
>
> This reverts commit a17651f7d8.
>
> Reason for revert: triggers failure in downstream test
>
> Original change's description:
> > Fix overflow due to rounding in AbsoluteSendTime::To24Bits
> >
> > Actual rounding is not important for this time as long it is consistent
> > during the call: only difference between two absolute send time matter
> > Rounding down avoids producing 1 < 24 when value is close to the wrap around boundary.
> >
> > Bug: None
> > Change-Id: Ibbf5bae21bc37eccdc5d4c130a59796ee5108017
> > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/268001
> > Commit-Queue: Åsa Persson <asapersson@webrtc.org>
> > Auto-Submit: Danil Chapovalov <danilchap@webrtc.org>
> > Reviewed-by: Åsa Persson <asapersson@webrtc.org>
> > Cr-Commit-Position: refs/heads/main@{#37468}
>
> Bug: None
> Change-Id: I90a9c1b174b918b7ede58c3bbdb879b1b67da7b2
> No-Presubmit: true
> No-Tree-Checks: true
> No-Try: true
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/268120
> Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
> Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
> Auto-Submit: Danil Chapovalov <danilchap@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#37473}
Bug: None
Change-Id: I99bcc6c6b7c08cd9621bdce336cd5793f78ee657
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/268190
Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37498}
This reverts commit a17651f7d8.
Reason for revert: triggers failure in downstream test
Original change's description:
> Fix overflow due to rounding in AbsoluteSendTime::To24Bits
>
> Actual rounding is not important for this time as long it is consistent
> during the call: only difference between two absolute send time matter
> Rounding down avoids producing 1 < 24 when value is close to the wrap around boundary.
>
> Bug: None
> Change-Id: Ibbf5bae21bc37eccdc5d4c130a59796ee5108017
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/268001
> Commit-Queue: Åsa Persson <asapersson@webrtc.org>
> Auto-Submit: Danil Chapovalov <danilchap@webrtc.org>
> Reviewed-by: Åsa Persson <asapersson@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#37468}
Bug: None
Change-Id: I90a9c1b174b918b7ede58c3bbdb879b1b67da7b2
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/268120
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
Auto-Submit: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37473}
This reverts commit dd32562f24.
Reason for revert: Updated the original change to dynamically load
the CoreMessaging.dll instead of statically linking with the .lib.
Original change's description:
> Revert "Wait for frames to arrive in WgcCapturer instead of returning nothing."
>
> This reverts commit 93bb305149.
>
> Reason for revert: It breaks a test while rolling into Chromium,
> see https://webrtc-review.googlesource.com/c/src/+/261780/21#message-4a96e33bfb475f19a618be82bbe72951b23085ef for details.
>
> Original change's description:
> > Wait for frames to arrive in WgcCapturer instead of returning nothing.
> >
> > We're seeing a high instance of "first capture failed" in Chromium when
> > using WGC. We can reduce this by waiting for frames to arrive if there
> > are none in the frame pool instead of returning a temporary error.
> >
> > I've set the maximum time to wait for a frame to 50ms. If no frame
> > arrives before 50ms has elapsed, we will return a temporary error.
> > Added a new test, FirstCaptureSucceeds, to verify that this is working
> > as expected.
> >
> > As part of this I updated the name of the `kCreateFreeThreadedFailed`
> > enum value to `kCreateFramePoolFailed`. The value remains the same
> > since they both report failures in frame pool creation.
> >
> > I also increased `kNumBuffers` from 1 to 2, so that the frame pool can
> > store two frames. This should prevent us from having to wait on the
> > event as frequently. This will increase the latency between capture
> > and display, however. High frame rate applications should not be
> > noticeably affected.
> >
> > Additionally, we uncovered a bug in the OS that prevents window capture
> > when there are displays attached, but none of them are active. Added
> > a new check to `IsWgcSupported` to cover this scenario.
> >
> > Finally, some issues with other WGC tests blocked moving the TryBots
> > to a newer version of Windows. This CL fixes those issues and updates
> > the TryBot configuration.
> >
> > bug: chromium:1314868
> > Change-Id: Id9c4d5ee98621e682ef04864c3848d50e761cdb7
> > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/261780
> > Reviewed-by: Alexander Cooper <alcooper@chromium.org>
> > Reviewed-by: Jeremy Leconte <jleconte@webrtc.org>
> > Commit-Queue: Austin Orion <auorion@microsoft.com>
> > Reviewed-by: Jeremy Leconte <jleconte@google.com>
> > Cr-Commit-Position: refs/heads/main@{#37404}
>
> Change-Id: If237df4826fe20b6fe2ca4b57253623321bf33c5
> No-Presubmit: true
> No-Tree-Checks: true
> No-Try: true
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/267460
> Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
> Owners-Override: Mirko Bonadei <mbonadei@webrtc.org>
> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
> Auto-Submit: Mirko Bonadei <mbonadei@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#37408}
Change-Id: I6cc2becd9ed363782ab2f326f58d9401bc8fb820
Bug: chromium:1314868
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/267902
Reviewed-by: Jeremy Leconte <jleconte@webrtc.org>
Reviewed-by: Alexander Cooper <alcooper@chromium.org>
Commit-Queue: Austin Orion <auorion@microsoft.com>
Cr-Commit-Position: refs/heads/main@{#37470}
Actual rounding is not important for this time as long it is consistent
during the call: only difference between two absolute send time matter
Rounding down avoids producing 1 < 24 when value is close to the wrap around boundary.
Bug: None
Change-Id: Ibbf5bae21bc37eccdc5d4c130a59796ee5108017
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/268001
Commit-Queue: Åsa Persson <asapersson@webrtc.org>
Auto-Submit: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37468}
https://webrtc-review.googlesource.com/c/src/+/250141 introduced a bug
due to which the min mic level override is always enforced, if specified
even if the user manually adjusts the mic level to zero.
This CL fixes that bug, the changes run behind a kill switch.
TESTED=Test video call on Chromium on Mac; input volume not adjusted after zeroing it from the system preferences UI
Bug: chromium:1275566
Change-Id: I18ce2e5970d3002b301f51f84544583c64982d57
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/267844
Reviewed-by: Hanna Silen <silen@webrtc.org>
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37460}
This CL removes the `WebRTC-Audio-AgcMinMicLevelExperiment` field trial
and always enables the code path behind that flag on Mac. In summary,
the analog AGC behaves as follows on Mac:
1. the minimum level is overridden to 20
2. the minimum is applied even when clipping is detected
3. when the level is manually adjusted to 0, the minimum level is
enforced - i.e., 20
Note that the 3rd property had been unintentionally added when the
changes were added behind the aforementioned field trial. This will
be fixed in a follow-up CL.
Bug: chromium:1275566
Change-Id: If184c4455a0780fcd94f55141af34460c152e3c3
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/266488
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Reviewed-by: Hanna Silen <silen@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37459}
Also move ScalabilityModeToString to api and add RTC_EXPORT so that
Chromium can use it.
Bug: chromium:986069
Change-Id: I5dbbb6de9b14ca20f3ae0630552dcd44595ad5ef
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/267780
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Daniel.L (Byoungchan) Lee <daniel.l@hpcnt.com>
Cr-Commit-Position: refs/heads/main@{#37444}
This CL also removes the existing non-standard implementation of the metric.
Bug: webrtc:14147, webrtc:11789
Change-Id: I70fd1c451dfd59380fe5ce959086f37b31697c16
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/265360
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Commit-Queue: Ivo Creusen <ivoc@webrtc.org>
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37441}
This CL duplicates a few lines of utility code from
//modules/audio_processing:audioproc_test_utils (which contains more
testonly things) and allows the possibility to remove testonly from
the unpack_aecdump tool.
Bug: b/237526033
Change-Id: If2e1dd4cc825429c496091cf8640c67069fb6e6f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/267701
Reviewed-by: Per Åhgren <peah@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37437}
