Notifier is thread-hostile, and we have added a SequenceChecker
on https://webrtc-review.googlesource.com/c/src/+/252520 ,
so this comment is no longer needed.
Bug: None
Change-Id: I39f7f75a786dd27d2f6299d85676e7182d9032eb
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/288180
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Auto-Submit: Daniel.L (Byoungchan) Lee <daniel.l@hpcnt.com>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38899}
In order to experiment with AGC2 and TS at the same time, 3 field
trials are removed and merged into `WebRTC-Audio-GainController2`,
which is existing.
New parameters for the `WebRTC-Audio-GainController2` field trial:
- `switch_to_agc2`: true by default; when true, the gain control
switches to AGC2 (both for the input volume and for the adaptive
digital gain);
- `min_input_volume`: minimum input volume enforced by the input
volume controller when the applied input volume is not zero;
- `disallow_transient_suppressor_usage`: when true, TS is never
created.
Removed field trials:
- `WebRTC-Audio-Agc2-MinInputVolume`: now a parameter of
`WebRTC-Audio-GainController2`;
- `WebRTC-ApmTransientSuppressorKillSwitch`: now a parameter of
`WebRTC-Audio-GainController2`;
- `WebRTC-Audio-TransientSuppressorVadMode`: automatically inferred
from `WebRTC-Audio-GainController2`.
Bug: webrtc:7494
Change-Id: I452798c0862d71f9adae6d163fe841df05ca44d6
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/287861
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Reviewed-by: Hanna Silen <silen@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38890}
In InputVolumeController, rename AnalyzePreProcess() and Process() to
reflect their use and replace the use of the getter
recommended_input_volume() with an optional return value from the
latter one. The added return value carries the recommended input
volume if the call sequence follows the API contract. Make the member
applied_input_volume_ optional. Restrict the use of the getter
recommended_input_volume() for test use. Add a method
capture_output_used() for test use.
In GainController2, store the output of InputVolumeController::Process()
in a new member variable that's updated in Analyze() and Process(). Use
a trivial getter to read the value in APM.
Bug: webrtc:7494
Change-Id: Ifcfb466c4f558be560eb6d2f45410d04adb7e2ad
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/287862
Reviewed-by: Alessio Bazzica <alessiob@webrtc.org>
Commit-Queue: Hanna Silen <silen@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38889}
The implementation here has a number of changes that force the callers
that called the "channel" functions into specific interfaces rather than
just letting C++ take care of it; this should go away once there stops
being a common implementation class for those interfaces.
Bug: webrtc:13931
Change-Id: Ic4e279528a341bc0a0e88d2e1e76c90bc43a1035
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/287640
Reviewed-by: Florent Castelli <orphis@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38888}
showing whether the DTLS transport is active and will send
a client hello or not.
BUG=None
Change-Id: If5c631c5e8701b0d1a4bb484467c93027b994ed7
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/287863
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Philipp Hancke <phancke@microsoft.com>
Cr-Commit-Position: refs/heads/main@{#38886}
This CL reverts change that was made in https://webrtc-review.googlesource.com/c/src/+/174220 where capturing of frame rendered time was moved into VideoFrameMetaData and construction of VideoFrameMetaData was placed before renderer->OnFrame(). That change made video freeze metrics (freeze and pause durations, harmonic frame rate) more off from the actual user experience.
Bug: webrtc:11489, b/261512902
Change-Id: Ic92a0cc1bb6d7b3ee1023804a73aa282848f8c4b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/287880
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38883}
This CL adds functionality to remove packets matching a given SSRC from
the pacer queue, and calls that with any SSRCs used by an RTP module
when that module is removed.
Bug: chromium:1395081
Change-Id: I13c0285ddca600e784ad04a806727a508ede6dcc
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/287124
Reviewed-by: Jakob Ivarsson <jakobi@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38880}
Only allow the application of min input volume if the applied input
volume is above zero. To implement this, add a member variable to
store the applied input volume. Rename the related setter to reflect
its new functionality.
Bug: webrtc:7494
Change-Id: Ia70d5cb4dfd972aad9ef2663a81884f3e5cb0758
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/287680
Reviewed-by: Alessio Bazzica <alessiob@webrtc.org>
Commit-Queue: Hanna Silen <silen@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38878}
Add an override to TurnPort for SendBindingErrorResponse to check
if a matching connection object exists before continuing. This is
needed to match with the check in `TurnPort::DispatchPacket` whereby
we forward calls to `Port` when no matching connections exist.
Bug: chromium:1395625
Change-Id: Idf1f898c2a93de6bc2832268db1cadd52cae23a9
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/287223
Reviewed-by: Sameer Vijaykar <samvi@google.com>
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38871}
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}
Follow-up of https://webrtc-review.googlesource.com/c/src/+/287221,
instead of asking GTest for the test suite and the test name, let's
propagate the test case passed by the user of PCLF.
Bug: b/237982523, webrtc:14757
Change-Id: Ia2a6ed4781f8c53c25b0006b8c7483e08ecead26
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/287126
Auto-Submit: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38867}
increasing the line coverage to 100%.
Note that code coverage is still a very basic metric and
the tests still do not enforce the length of the attributes
nor their token-char-ness defined in
https://www.rfc-editor.org/rfc/rfc8830.html#section-2
BUG=webrtc:14745
Change-Id: If5de5e1f8c32f7dae029916c6cd0a4f2b094f672
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/286900
Commit-Queue: Philipp Hancke <phancke@microsoft.com>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Florent Castelli <orphis@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38864}
Pass the correct number of channels needed by the AGC2 input volume
controller. This change doesn't affect the adaptive digital
controller which reads the number of channel from the passed audio
buffer instance for each processed frame.
Note that the `AdaptiveDigitalGainController::Initialize()` impl
was removed in [1], but that CL didn't remove the declaration (done
in this CL).
[1] https://webrtc-review.googlesource.com/c/src/+/287222/5/modules/audio_processing/agc2/adaptive_digital_gain_controller.cc#105
Bug: webrtc:7494
Change-Id: I07369ab4025a251b25c716cf618e4222fdb60fc8
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/287320
Reviewed-by: Hanna Silen <silen@webrtc.org>
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38863}
When running Gtest-based tests, we use the WebRtcUnitTestDelegate
to call Gtest from the iOS application.
However, if tests take too long, iOS terminates that application.
The test execution code (ios/build/bots/scripts) retries up to three times,
but that functionality doesn't seem to work lately.
Because of this, module_unittests are failing at a fairly high rate.
Instead, use GoogleTestRunner to let XCTest run Gtest-based tests.
This is enabled with the --enable-run-ios-unittests-with-xctest flag,
which is passed when using the --xctest flag in ios/build/bots/scripts/run.py.
Existing XCTest-based tests (eg. sdk_unittest) are not affected
as the --xcode-parallelization flag takes precedence over --xctest.
Bug: None
Change-Id: Ib7f8a6d24f6b25444a47e3a83c0edbe96318be46
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/287180
Reviewed-by: Jeremy Leconte <jleconte@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Commit-Queue: Daniel.L (Byoungchan) Lee <daniel.l@hpcnt.com>
Cr-Commit-Position: refs/heads/main@{#38859}
This metadata key is temporary, as explained in bugs.webrtc.org/14757,
this information will be at some point directly accessible via the
webrtc.test_metrics.Metric.test_case field.
Bug: b/237982523, webrtc:14757
Change-Id: Ie77875a33db5961f8a5572bd1b7066ad8ba17291
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/287221
Reviewed-by: Jeremy Leconte <jleconte@google.com>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38858}
Add AGC2 digital adaptive config parameters in the field trial
"WebRTC-Audio-InputVolumeControllerExperiment". Rename it as
"WebRTC-Audio-GainController2" to reflect that the override now adjusts
the parameters for both input volume controller and adaptive digital
controller.
Bug: webrtc:7494
Change-Id: Ifbc1b8be76cf23b0b6b74b22b5167a45972cab38
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/286880
Reviewed-by: Alessio Bazzica <alessiob@webrtc.org>
Commit-Queue: Hanna Silen <silen@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38855}
Rename MonoInputVolumeController member input_volume_ to reflect its
use to store the most recent input volume recommendation.
Rename the remaining variables named as manager in the unit tests.
Bug: webrtc:7494
Change-Id: I31ffdc131c98061ef2b36f98b685c5182b3c6861
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/287123
Reviewed-by: Alessio Bazzica <alessiob@webrtc.org>
Commit-Queue: Hanna Silen <silen@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38854}
The `WebRTC.Audio.AgcSetLevel` name is misleading and the histogram
is logged for each channel - but the input volume is one for all the
channels.
Changes:
- `WebRTC.Audio.Apm.RecommendedInputVolume.OnChangeToMatchTarget`
is the new name
- Now available not only in `AgcManagerDirect` (AGC1), but also in
`InputVolumeController` (AGC2)
- Logged once and not for each channel
- Also add the following AGC implementation agnostic histograms
- `WebRTC.Audio.Apm.AppliedInputVolume.OnChange`
- `WebRTC.Audio.Apm.RecommendedInputVolume.OnChange`
- Fix `SpeechSamplesReader::Feed()` in the unit tests, which did
not set the applied input volume and apply the recommended one
The histogram definitions are updated in crrev.com/c/4087426.
Bug: webrtc:7494
Change-Id: I03c5dfb08165805215ca2c4bb6509b16de8d68da
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/287081
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Reviewed-by: Hanna Silen <silen@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38852}
This will enable loss based bwe v2 by default. The default params were used in Chrome experiment and got positive result. Remove some tests in goog_cc, which are for loss based bwe v1.
Bug: webrtc:12707
Change-Id: Ice126a128f6e8cea8b861f879d09e390ee69e521
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/285740
Commit-Queue: Diep Bui <diepbp@google.com>
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38851}
These histograms have expired and have no owner.
Remove to clean up the code and save memory.
Fixed: chromium:1117100
Change-Id: I24a009d8e432109c1d62c4a3a16eff5cd21c8541
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/286660
Commit-Queue: Johannes Kron <kron@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38850}
