Replace kUpdateInputVolumeWaitFrames with
update_input_volume_wait_frames in InputVolumeController::Config.
Also, fix an off-by-one error in the frame count to give a better
readability for non-zero wait frames. Now
update_input_volume_wait_frames_ = 100 allows updates every 100 frames
instead of every 101 frames. Effectively, this makes
update_input_volume_wait_frames = 0 and 1 to behave similarly (i.e.,
they now both allow updates after every frame).
Bug: webrtc:7494
Change-Id: I597f7e88895a4dcd365dc6dee526acb9d971b2fc
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/282863
Reviewed-by: Alessio Bazzica <alessiob@webrtc.org>
Commit-Queue: Hanna Silen <silen@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38648}
Add loss_limited_probe_scale as a scale factor which decides how much we should probe when bandwidth is loss limited.
Bug: webrtc:12707
Change-Id: I194b2b40c9a7861d82b61585bcaf484ab228eedb
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/281360
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Commit-Queue: Diep Bui <diepbp@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38636}
Motivation: loss based ramp-up can be incorrect when (1) bandwidth is loss limited, and (2) delay based estimate might be incorrect due to no delay signals. Therefore, bounding the loss based estimate by the delay based estimate is not much helpful in those cases.
Thus strengthening the bounding logic by using upper link capacity is one of solutions to avoid incorrect ramp-up.
Without the change: screen/qmLedxapJWvUTmn
With the change: screen/8sQcksWa6CptywK
Bug: webrtc:12707
Change-Id: I32ba82693b3ffa83cbb89c2cc9690fe16fb10c78
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/283085
Commit-Queue: Diep Bui <diepbp@webrtc.org>
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38626}
Openh264 switched from api/svc to api/wels as the location for some
codec header files. During the transition it was necessary to
conditionally from either the old or new location, but now that the
switch is completed and has settled for about two weeks the conditionals
can be removed. This finishes the #include transition started by
webrtc-review.googlesource.com/c/280800
Bug: chromium:1218384
Change-Id: Ic0847428d134687908cc26fec1fdec0c612674b2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/281561
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Auto-Submit: Bruce Dawson <brucedawson@chromium.org>
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38622}
BurstyPacer is currently controlled via field trials. In order for
Chrome to be able to have burst without relying on a field trial, this
parameter is added.
When all burst experiments have concluded we may be able to have a
hardcoded constant instead, but for now the parameter is added to
RTCConfiguration.
NOTRY=True
Bug: chromium:1354491
Change-Id: I386c1651dbbcbf309c15ea3d3380cf8f632b5429
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/283420
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38621}
This is a reland of commit e6ec81a89ca904f1816b76456426babc28a9d767
Updated to ensure that the portal code can be built with is_chromeos.
Original change's description:
> Split out generic portal / pipewire code
>
> It will be reused by the video capture portal / pipewire backend.
>
> Bug: webrtc:13177
> Change-Id: Ia1a77f1c6e289149cd8a1d54b550754bf192e62e
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/263721
> Reviewed-by: Mark Foltz <mfoltz@chromium.org>
> Commit-Queue: Alexander Cooper <alcooper@chromium.org>
> Reviewed-by: Erik Språng <sprang@webrtc.org>
> Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
> Reviewed-by: Salman Malik <salmanmalik@google.com>
> Cr-Commit-Position: refs/heads/main@{#38487}
Bug: webrtc:13177
Change-Id: I2c890c83c86ad60fa30f63dcf6fa90510d46009e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/281661
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Reviewed-by: Alexander Cooper <alcooper@chromium.org>
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Commit-Queue: Alexander Cooper <alcooper@chromium.org>
Cr-Commit-Position: refs/heads/main@{#38620}
Patchset 1 contrains the original cl.
Later patchsets contain fix.
Original description:
Continue probing if networkstat estimate increase
This fixes an issue where continues probing stops if networkstate estimate is low when a probe is sent, but increase as a consequence of the probe.
Bug: webrtc:14392
Change-Id: I8d4e1968020f9f8de18e12a4a0322a87f1a8fd2f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/283082
Reviewed-by: Diep Bui <diepbp@webrtc.org>
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38612}
This reverts commit dd7dc25a30c5841e6620d195b83058a22ffff7cd.
Reason for revert: Bug in CL. Continuously probe if experiment for probing based on the link capacity is enabled.
Original change's description:
> Continue probing if networkstat estimate increase
>
> This fixes an issue where continues probing stops if networkstate estimate is low when a probe is sent, but increase as a consequence of the probe.
>
>
> Bug: webrtc:14392
> Change-Id: Id1d703f7efc824a6a6f8d899c367660291bd88c8
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/282941
> Reviewed-by: Diep Bui <diepbp@webrtc.org>
> Commit-Queue: Per Kjellander <perkj@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#38606}
Bug: webrtc:14392
Change-Id: Ib241b190951a78c436188c0b83d0247bf7d0dddd
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/283080
Auto-Submit: Per Kjellander <perkj@webrtc.org>
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com>
Cr-Commit-Position: refs/heads/main@{#38609}
This fixes an issue where continues probing stops if networkstate estimate is low when a probe is sent, but increase as a consequence of the probe.
Bug: webrtc:14392
Change-Id: Id1d703f7efc824a6a6f8d899c367660291bd88c8
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/282941
Reviewed-by: Diep Bui <diepbp@webrtc.org>
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38606}
This is to avoid passing delay based estimate value twice from send side bwe.
Bug: webrtc:12707
Change-Id: Idc77cf7c2f4ecc60ae1dcfead325320532e7a7ca
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/282864
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Commit-Queue: Diep Bui <diepbp@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38600}
It appears to be still failing occasionally so add one more event
to verify streams connected successfully in order to verify whether
we sent and received buffers properly in the next step.
Bug: webrtc:14644
Change-Id: I08822b15452fc845d68cbff1b01ae6b6f7c1f486
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/282842
Reviewed-by: Alexander Cooper <alcooper@chromium.org>
Reviewed-by: Jeremy Leconte <jleconte@google.com>
Commit-Queue: Jan Grulich <grulja@gmail.com>
Cr-Commit-Position: refs/heads/main@{#38598}
Instead of disabling probing when the total allocated bitrate has
changed in goog_cc, it can be done via a new field trial parameter,
"probe_max_allocation". Not that the currently used flag
RateControlSettings::TriggerProbeOnMaxAllocatedBitrateChange() is per
default enabled and will be cleaned up in a follow up cl.
The field trial flag "skip_if_est_larger_than_fraction_of_max" now also
skip probing if the current estimate is larger than the currently max
allocated bitrate. ie, alr probing is skippe if the current estimate >
max configured bitrate or current estimate > max send bitrate of all
streams.
Bug: webrtc:14392
Change-Id: I2a09be39f85a9122410edd5acb1158ece12fca60
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/282860
Reviewed-by: Diep Bui <diepbp@webrtc.org>
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38597}
Replace the use of MonoInputController::min_mic_level() with
MonoInputVolumeController::clipped_level_min() when estimating input
volume adjustment from clipping prediction. The adjustment is later
capped in MonoInputVolumeController::HandleClipping() using
clipped_level_min_ so no audio changes are expected from this change.
Bug: webrtc:7494
Change-Id: Ie26d0aa5cce3eeef06f70a281504889519bb5aca
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/282840
Reviewed-by: Alessio Bazzica <alessiob@webrtc.org>
Commit-Queue: Hanna Silen <silen@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38593}
Instead of trying to guess the state from the loss based estimator by
looking at the estimate, use the state.
Bug: webrtc:14392
Change-Id: Ibf6e762f02bfbfff175f2aa2bc98f45b1c5beb1a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/282823
Reviewed-by: Diep Bui <diepbp@webrtc.org>
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38589}
Previously, cleanup in GetCandidateBandwidthUpperBound in loss_based_bwe_v2.cc causes unbounded bandwidth estimate. It leads to many warning logs being printed out at loss_based_bwe_v2.cc:95.
However, the cleanup is still necessary because the param bandwidth_rampup_upper_bound_factor is not used in current launches.
To fix the infinite estimate, we set max_bitrate in loss based bwe, which is derived from goog_cc, and not allow the estimate to go above that value.
*** Original change description ***
* Revert "Probing integration in loss based bwe 2." (diepbp@webrtc.org)
* https://webrtc-review.googlesource.com/c/src/+/277400
***
Bug: webrtc:12707
Change-Id: If0cd16daba4a4941043a1610edca2a13c9564328
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/281280
Commit-Queue: Diep Bui <diepbp@webrtc.org>
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38574}
This =default declaration has no effect other than to break designated
initialization in C++20 by making the type no longer an aggregate.
Bug: None
Change-Id: I20a4c285b7cbfed074291b9ee27c03aa29bada32
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/281960
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Taylor Brandstetter <deadbeef@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38573}
This is a reland of commit 38fcd58429b29c9474f1647efed7ebeb543c0637
Original change's description:
> Change default NetEq sample rate to 48k.
>
> This should avoid some resampling before any packets have been received given that the vast majority of devices use 48k sample rate and the most common codec is Opus (which we always decode in 48k).
>
> Bug: none
> Change-Id: Ie7baea57c3eb1b763a6460c3b06b56d67b2b258e
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/280662
> Commit-Queue: Jakob Ivarsson <jakobi@webrtc.org>
> Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#38536}
Bug: none
Change-Id: Id634799286f6d1f1eaf315ebe8e70de669d589db
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/281900
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Commit-Queue: Jakob Ivarsson <jakobi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38572}
Remove function declarations, members, and friend tests that are
no longer used. Reorder the member variables.
Bug: webrtc:7494
Change-Id: I8c24e2f4b9d9846e6d3fef4e2c998aa26f49f8c9
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/282180
Reviewed-by: Alessio Bazzica <alessiob@webrtc.org>
Commit-Queue: Hanna Silen <silen@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38570}
Replace the use of speech level target and digital gain maximum with speech level target range parameters.
Bug: webrtc:7494
Change-Id: I703756c5a3fbd330ed585e3f5b4ac3141d9ea6e2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/280943
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Reviewed-by: Alessio Bazzica <alessiob@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38563}
This is a reland of commit c1d5fda22c8ae456950c5549d22d099b478c67e2
Original change's description:
> Add documentation, tests and simplify webrtc::SimulatedNetwork.
>
> This CL increases the test coverage for webrtc::SimualtedNetwork, adds
> some more comments to the class and the interface it implements and
> simplify the logic around capacity and delay management in the
> simulated network.
>
> More CLs will follow to continue the refactoring but this is the
> ground work to make this more modular in the future.
>
> Bug: webrtc:14525, b/243202138
> Change-Id: Ib0408cf6e2c1cdceb71f8bec3202d2960c5b4d3c
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/278042
> Reviewed-by: Artem Titov <titovartem@webrtc.org>
> Reviewed-by: Per Kjellander <perkj@webrtc.org>
> Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
> Reviewed-by: Björn Terelius <terelius@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#38388}
Bug: webrtc:14525, b/243202138, b/256595485
Change-Id: Iaf8160eb8f8e29034b8f98e81ce07eb608663d30
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/280963
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38557}
In InputVolumeController/MonoInputVolumeController, set
min_digital_gain_db_ and disable_digital_adaptive_ to fixed values
ahead of replacing speech level target as well as digital gain
minimum and maximum with target range parameters.
In InputVolumeController, remove digital_adaptive_follows and
min_digital_gain_db from the config as they are no longer needed.
Bug: webrtc:7494
Change-Id: I1378b6e182224c41038c6d8c649e7a28961f73d4
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/280962
Reviewed-by: Alessio Bazzica <alessiob@webrtc.org>
Commit-Queue: Hanna Silen <silen@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38554}
Modify unit tests ahead of changes that will replace the minimum
digital gain with a fixed value 0 and always enable digital gain
compensation.
Bug: webrtc:7494
Change-Id: I9df95667b831d5b68e70aaba22f631b398edf8e2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/280960
Commit-Queue: Hanna Silen <silen@webrtc.org>
Reviewed-by: Alessio Bazzica <alessiob@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38553}
Rename constants and arguments reflecting the old naming with RMS error
overriding the error calculated by the analog AGC. Rename the related
unit tests and helper functions.
Bug: webrtc:7494
Change-Id: I9a1d972e9ff7ab5cdd43ca3568379d511801adee
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/280481
Reviewed-by: Alessio Bazzica <alessiob@webrtc.org>
Commit-Queue: Hanna Silen <silen@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38552}
Add InputVolumeController as a member in GainController2 (not created
by default). Add a method GainController2::Analyze() to update the
applied input volume and run the pre-processing steps in
InputVolumeController. Add a call InputVolumeController::Process() in
GainController2::Process().
Bug: webrtc:7494
Change-Id: Idf4111ac5e19a620b6421c7f23fd642f169c7b5a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/279822
Reviewed-by: Per Åhgren <peah@webrtc.org>
Reviewed-by: Alessio Bazzica <alessiob@webrtc.org>
Commit-Queue: Hanna Silen <silen@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38548}
This should avoid some resampling before any packets have been received given that the vast majority of devices use 48k sample rate and the most common codec is Opus (which we always decode in 48k).
Bug: none
Change-Id: Ie7baea57c3eb1b763a6460c3b06b56d67b2b258e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/280662
Commit-Queue: Jakob Ivarsson <jakobi@webrtc.org>
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38536}
clarifying that the number of temporal layers is limited to
a single byte and moving the format description from the source
to the document.
drive-by editorial fixes
BUG=webrtc:12000
Change-Id: I33f85e0a81e1dc16ef762171c52a79919080e048
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/279940
Commit-Queue: Philipp Hancke <phancke@microsoft.com>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38523}
The clang pragma have been added to ensure we can still test the code
until usage is gone, and that we can still have the one implementation
compiling without itself tripping on the deprecation errors.
Users of the code will have deprecation warnings or error as intended.
Bug: webrtc:14617
Change-Id: I21dae57c669557d4d218c235c811174a477be080
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/281221
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Commit-Queue: Florent Castelli <orphis@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38514}
Copy AgcManagerDirect files from agc to agc2. Rename the newly
created files and classes ahead of refactoring. Add a build
target.
This change is done to enable creating a class
InputVolumeController based on AgcManagerDirect. The added
temporary dependency on files in agc will be removed
in https://webrtc-review.googlesource.com/c/src/+/278625.
The exact copy of the files happened in the 1st patchset and it
has been verified as follows:
Checksum check:
```
$ git checkout main && git pull
# Go back to the tree state before [1] landed
$ git new-branch tmp
$ git reset --hard 2235776597e2f47ec353ac911428eb9a54d64a10
$ cd modules/audio_processing/agc/
$ md5 agc_manager_direct*
MD5 (agc_manager_direct.cc) = e661481a85f72596cae4599b62907f5b
MD5 (agc_manager_direct.h) = bf68280e2d0f689b4ebcd665b5db6052
MD5 (agc_manager_direct_unittest.cc) = 6bf0bf45ff5e940b1a3bb37154f09269
```
Patchset 1 (see [2])
```
$ cd modules/audio_processing/agc2/
$ md5 input_volume_controlle*
MD5 (input_volume_controller.cc) = e661481a85f72596cae4599b62907f5b
MD5 (input_volume_controller.h) = bf68280e2d0f689b4ebcd665b5db6052
MD5 (input_volume_controller_unittest.cc) = 6bf0bf45ff5e940b1a3bb37154f09269
```
[1] https://webrtc-review.googlesource.com/c/src/+/278781
[2] https://webrtc-review.googlesource.com/c/src/+/278624/1
Bug: webrtc:7494
Change-Id: I7804da899d18adf556b089c76a567ce27c299a62
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/278624
Reviewed-by: Alessio Bazzica <alessiob@webrtc.org>
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Commit-Queue: Hanna Silen <silen@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38512}
This doesn't effect for how long the test will run, it just gives
PipeWire more time to establish connection and create empty buffers
before we try to work with it. All the waiting events will be
interrupted by signals once we no longer need to wait so it doesn't
matter if we wait 2 seconds or 5 seconds.
Bug: webrtc:14568
Change-Id: Ie918e8943bf882059b1289f57595fc302216745e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/280700
Reviewed-by: Alexander Cooper <alcooper@chromium.org>
Commit-Queue: Jan Grulich <grulja@gmail.com>
Cr-Commit-Position: refs/heads/main@{#38486}
