Makes construction simpler, and allows the ts_extrapolator_ pointer
to be marked const.
Followup to https://webrtc-review.googlesource.com/c/src/+/190721
Bug: webrtc:12102
Change-Id: I2abeb960935b5470509f654a4a9d5121c8001900
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/190981
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32535}
Avoid excessive echo suppression in frequencies above 2 kHz when
there is a dominant nearend. Calls with clock drift will not be affected
by this change as they tend to have less accurate linear filters.
Bug: webrtc:11985
Change-Id: Iddc628da5e2ba572c1b47acd87dd3be35260dca1
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/188580
Reviewed-by: Per Åhgren <peah@webrtc.org>
Commit-Queue: Gustaf Ullberg <gustaf@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32533}
After recently changing .pylintrc (see [1]) we discovered that
the presubmit check always checks all the python files when just
one python file gets updated.
This CL moves all these files one step closer to what the linter
wants.
Autogenerated with:
# Added all the files under pylint control to ~/Desktop/to-reformat
cat ~/Desktop/to-reformat | xargs sed -i '1i\\'
git cl format --python --full
This is part 1 out of 2. The second part will fix function names and
will not be automated.
[1] - https://webrtc-review.googlesource.com/c/src/+/186664
No-Presubmit: True
Bug: webrtc:12114
Change-Id: Idfec4d759f209a2090440d0af2413a1ddc01b841
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/190980
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32530}
Only use, in VCMTiming, protects it with its own mutex.
Bug: webrtc:12102
Change-Id: I9c09976f9d938565b3e2908eca6cfee0c4063f6f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/190721
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Reviewed-by: Tommi <tommi@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32529}
This change moves ScopedComInitializer out of core_audio_utility and
into rtc_base/win so it can be reused elsewhere more easily.
It also adds HSTRING and GetActivationFactory functionality to
rtc_base/win. These two were heavily based on what is already present
base/win.
All of these are necessary for the new window capturer based on the
Windows.Graphics.Capture API. You can see how these APIs will be
used in this CL: 186603: Implement WgcCaptureSession |
https://webrtc-review.googlesource.com/c/src/+/186603
Bug: webrtc:9273
Change-Id: I0a36373aac98be779ccbabe1053bb8d6e234f6a3
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/188523
Commit-Queue: Tommi <tommi@webrtc.org>
Reviewed-by: Tommi <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32522}
This CL includes two changes:
1. the type for (inverted) lags and pitch periods changed from
size_t to int to reduce the chance of bugs with pitch period
manipulations
2. CheckLowerPitchPeriodsAndComputePitchGain() is optimized by
replacing an unnecessary if statement inside the loop with the
predetermined number of loops
Bug: webrtc:10480
Change-Id: I38432699254b37a2c0111279c28be8dc65b87e9b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/139252
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Reviewed-by: Fredrik Hernqvist <fhernqvist@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32521}
Calculate quality metrics for dropped frames by comparing original
frame against last decoded one.
This feature makes comparison of encoders which do/don't drop frames
more fair.
The feature is controlled by analyze_quality_of_dropped_frames flag
and is disabled by default.
Bug: none
Change-Id: Ifab8df92d0b76e743ff3193c05d7c8dbd14921c4
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/190660
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32518}
Large gaps can cause issues in NetEq simulations, so the simulation is
ended whenever we encounter one. However, the time span of the gap is
still included in the simulation time, leading to incorrect results.
Bug: webrtc:10337
Change-Id: I94a1a0b46259e3718b1b73522a3886a17bedbb7d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/190287
Reviewed-by: Jakob Ivarsson <jakobi@webrtc.org>
Commit-Queue: Ivo Creusen <ivoc@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32514}
In the old jitter buffer the two VCMVideoProtection modes |kProtectionNone| and |kProtectionFEC| could be set on the jitter buffer for it to not wait for NACK and instead generate incomplete frames. This has not been possible for a long time.
Bug: webrtc:9378, webrtc:7408
Change-Id: I0a2d3ec34d721126c1128306d5fad88314f8d59f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/190680
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32513}
This code used to have a reader-writer lock, and call
std::queue::pop() with only a reader lock, which appears unsafe. Code
changed to use a plain webrtc::Mutex.
Bug: webrtc:12102
Change-Id: Icbea17a824c91975dfebd4d05bbd0c21e1abeadc
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/190700
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32511}
INCREASING_ID, which is the default mode, triggers HW reset in chromium
decoder wrapper. Set eSpsPpsIdStrategy to SPS_LISTING to prevent that.
Note that WebRTC always resets the encoder on resolution change. This
makes all strategies except INCREASING_ID essentially equivalent to
CONSTANT_ID.
Bug: chromium:1111273
Change-Id: I37405c97b3390f812d1dcaa111694b3b1d638035
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/190440
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32505}
"warning: control reaches end of non-void function [-Wreturn-type]"
Reported by gcc (8.3)
In all the reported cases, the end of function is never actually
reached. Add RTC_CHECK(false) to ensure the compiler is aware that
this path is a dead-end.
Bug: webrtc:12008
Change-Id: I7f816fde3d1897ed2774057c7e05da66e1895e60
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/189784
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Fabien VALLÉE <fabien.vallee@netgem.com>
Cr-Commit-Position: refs/heads/master@{#32503}
This feature is active if and only if the RTP header extension
playout-delay is used with min playout delay=0 and max playout delay>0.
In this case, a maximum composition delay will be calculated and attached
to the video frame as a signal to use the low-latency renderer algorithm,
which is landed in a separate CL in Chromium.
The maximum composition delay is specified in number of frames and is
calculated based on the max playout delay.
The feature can be completetly disabled by specifying the field trial
WebRTC-LowLatencyRenderer/enabled:false/
Bug: chromium:1138888
Change-Id: I05f461982d0632bd6e09e5d7ec1a8985dccdc61b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/190141
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Johannes Kron <kron@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32493}
This CL removes the circular shift to compute the convolution in
`ComputeLpResidual()`, which is now 4x faster (benchmarked with the
`RnnVadTest.DISABLED_ComputeLpResidualBenchmark` unit test).
Note that the `RnnVadTest.LpResidualPipelineBitExactness` unit test
is still passing.
Bug: webrtc:10480
Change-Id: Ia7767d9b57378c12c8ff31f58fea03905be5c5de
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/189964
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Reviewed-by: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32491}
Use hard-coded constants previously computed at runtime within a loop in
`DenoiseAutoCorrelation`.
Note that the `RnnVadTest.LpResidualPipelineBitExactness` unit test
is still passing.
Bug: webrtc:10480
Change-Id: I02c2fff7dc7153ea2ab8a27cad8a479a0f029996
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/189963
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Reviewed-by: Jakob Ivarsson <jakobi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32489}
This changes the default behavior to use pacing factor of 1.1x instead
of 2.5x, it also sets libvpx rate controler as trusted, turns on the
encoder pushback mechanism and sets spatial hysteresis to 1.2.
The unused "dynamic rate" settings in libvpx is removed.
The new settings matches what has been used in chromium since 2019.
If needed, the legacy behavior can be enabled using the field trial
WebRTC-VideoRateControl.
Bug: webrtc:10155
Change-Id: I8186b491aa5bef61e8f568e96c980ca68f0c208f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/186661
Commit-Queue: Erik Språng <sprang@webrtc.org>
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Christoffer Rodbro <crodbro@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32477}
The cross-correlation function in the anonimous namespace of
lp_residual.cc is used once as to compute auto-correlation coefficients.
Hence, it can be simplified (and renamed).
Note that the `RnnVadTest.LpResidualPipelineBitExactness` unit test
is still passing.
Bug: webrtc:10480
Change-Id: Ie769e42907c15572172a8ad1a34e82ce60c4a386
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/189962
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32464}
Use hard-coded constants previously computed at runtime within a loop in
`ComputeAndPostProcessLpcCoefficients`.
Note that the `RnnVadTest.LpResidualPipelineBitExactness` unit test
is still passing.
Bug: webrtc:10480
Change-Id: Iaec178c56937449c3fe9af881354d19cc8c47c7d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/189961
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32462}
This change makes the transparent mode classifier consider filter
convergence also for microphone signals with very low volume. This is
in order to prevent entering transparent mode when there is low, but
audible, echo.
Furthermore, the suppression gain during transparent mode is restored
to avoid leaks when the echo is too low to be reliably detected by
filter convergence.
Bug: webrtc:10232, chromium:1140452
Change-Id: Idd4f40c4aee7c20baa444afaa5ec604eb65bcfd0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/189786
Reviewed-by: Per Åhgren <peah@webrtc.org>
Commit-Queue: Gustaf Ullberg <gustaf@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32461}
These metrics are no longer in use:
WebRTC.Audio.EchoCanceller.ActiveRender
WebRTC.Audio.EchoCanceller.ErleBand*
WebRTC.Audio.EchoCanceller.ErlBand*
Corresponding CL to mark obsolete in histograms.xml:
https://chromium-review.googlesource.com/c/chromium/src/+/2485059
Bug: chromium:1140067,chromium:1140071,chromium:1140073
Change-Id: Icb79b540f340a596c178f352363b30dc7b885251
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/189783
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Commit-Queue: Sam Zackrisson <saza@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32449}
- Add buffer level filter and delay manager mocks and make them
injectable for easier testing.
- Add a basic set of tests for simple cases and recently added features.
Bug: webrtc:10333
Change-Id: I8b6f73b8ad99ad6859ed1279086c0bd68b7687be
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/188623
Commit-Queue: Jakob Ivarsson <jakobi@webrtc.org>
Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32433}
The DxgiOutputDuplicator uses a vector<byte> to hold the rects
that have changed on the screen. It then iterates over the
vector to grab each rect and apply it to the updated_region.
There is vector resizing logic which checks the 'capacity' of
the vector and determines whether there is enough space for the
changed rect data. Often the 'capacity' and 'size' of the
vector are equal but that isn't always true. When the capacity
is greater than size, and the number of changed rects is high
enough, rect data will be written past the element pointed to
by (data() + size()) and this is the error that ASAN is warning
of.
The fix is to use size() instead of capacity() when determining
whether to resize the vector and as the buffer size we provide
to the Windows API. There are no other usages of this vector so
there aren't any problems caused by the size/capacity discrepancy
in the existing builds. The ASAN issue is worth fixing in case
someone comes along and decides to use the vector differently (e.g
rely on the size instead of capacity so some of the rects are
not counted).
Bug: chromium:1138446
Change-Id: I3041091423de889e0f8aabc56ece9466a3000b4f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/188900
Reviewed-by: Jamie Walch <jamiewalch@chromium.org>
Commit-Queue: Joe Downing <joedow@google.com>
Cr-Commit-Position: refs/heads/master@{#32425}
The desktop_capture code logs failed HRESULTs in several places,
the problem is that it tries to use a wchar* (when a char*
is required) for the error message and doesn't display the HRESULT
in hex. This makes the error logging less useful than it could be.
Example failure: error 08406B28, with code -2005270488
In this CL, I add a simple utility function to convert a _com_error
object to a std::string whic can be logged. With this change the
output looks like this (linebreak added for CL description):
Failed to capture frame: HRESULT: 0x887A0026,
Message: 'The keyed mutex was abandoned.'
I also adjusted the formatting of a few logging statements to be
more consistent (adding '<<' operators mostly).
Bug: webrtc:12051
Change-Id: I3e88ff6f2ff079fbe210626e1e89b2b053a742a9
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/188522
Reviewed-by: Jamie Walch <jamiewalch@chromium.org>
Commit-Queue: Joe Downing <joedow@google.com>
Cr-Commit-Position: refs/heads/master@{#32417}
To make it natural to reuse them for vp9
Bug: webrtc:11999
Change-Id: If2ef7ca16b8be96e0e03bb19211d9f5eb74b2d3d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/188620
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32414}
This CL also puts the arguments in a struct to allow for easier future additions.
Bug: webrtc:11005
Change-Id: I47bf664e7106b724eb1fc42299c42bbf022393ef
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/188385
Commit-Queue: Ivo Creusen <ivoc@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Jakob Ivarsson <jakobi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32409}
to make it a bit simpler
Bug: None
Change-Id: Ie6288594c5a1b8535007623032b422eefc716ca6
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/188460
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32405}
AGC2 is correctly (re)initialized when its config changes.
This CL also improves the `AudioProcessingImpl::ApplyConfig`
readability by defining operator!= also for the AGC1 config.
Bug: webrtc:7494
Change-Id: I62068de32c941e6b18d4618c656f569647042345
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/187120
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Reviewed-by: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32402}
The APM config to string mapping must be in one place (namely,
in `audio_processing.cc`). This CL moves the AGC2 config to string
impl to the right place.
This CL also updates `GainController2::Validate()` and adds the
missing unit tests for the parameters that have recently been added.
Stack buffer size in `AudioProcessing::Config::ToString()` increased
because of the extra params. Syntax near `multi_channel_capture` fixed.
Output string format verified with a JS linter.
Bug: webrtc:7494
Change-Id: I692e1549b7d40c970d88a14c8e83da16325fb54c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/187080
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Reviewed-by: Minyue Li <minyue@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32400}
This reverts commit 84524e6b196153e35910111f61c36489ec2bd1d7.
Reason for revert: https://webrtc-review.googlesource.com/c/src/+/160649/33#message-a83e8959e03a274642ca2ce1abb9ea0099f08097 + suspected in breaking WebRTC to Chromium rolls https://chromium-review.googlesource.com/c/chromium/src/+/2468139
Original change's description:
> Improve screen sharing with PipeWire on Wayland
>
> Currently, sharing a screen or a window on Wayland opens unnecessary
> preview dialog on Chromium side, which is then followed by a similar
> dialog on xdg-desktop-portal side. The Chromium dialog is useless on
> Wayland, as it doesn't show anything. This is because Chromium doesn't
> have access to screen content as in case of X11 session. To fix this, we
> want to avoid showing the preview dialog in case we find that we run on
> Wayland and only pick a screen or a window from the dialog that comes
> from xdg-desktop-portal.
>
> This patch splits BaseCapturerPipeWire class, moving portal related code
> into XdgPortalBase, which does all the DBus communication and which is
> supposed to be reused by BaseCapturerPipeWire when the user confirms
> the dialog from xdg-desktop-portal. The XdgPortalBase is extended to
> support multiple calls at once, where each call is identified by Id.
>
> Relevant change on Chromium side will be in a different review.
>
> Bug: chromium:682122
> Change-Id: I2bcd07d16a5eb3b902db63ea9a164c5bd39c23a0
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/187492
> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
> Reviewed-by: Tommi <tommi@webrtc.org>
> Commit-Queue: Tommi <tommi@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#32388}
TBR=mbonadei@webrtc.org,tommi@webrtc.org,sprang@webrtc.org,guidou@webrtc.org,mfoltz@chromium.org,grulja@gmail.com
Change-Id: I1a9748ced5626903b12214d677c7b8919c2ac385
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: chromium:682122
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/188380
Reviewed-by: Olga Sharonova <olka@webrtc.org>
Commit-Queue: Olga Sharonova <olka@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32398}
This adds a method to allow VideoLayersAllocation to be sent using the header extension RtpVideoLayersAllocationExtension.
Bug: webrtc:12000
Change-Id: Iafdc1e16911c57ca55d7cc0559a0b45774211e92
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/187495
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32397}
