The code has been running in Chrome since 2020 and ChromeOS since 2022 (https://crrev.com/c/3452884) without issues.
Bug: webrtc:11803
Change-Id: I0c572d362b1f52b4591c7790e11a87c1a1ad1a79
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/293342
Reviewed-by: Per Åhgren <peah@webrtc.org>
Commit-Queue: Sam Zackrisson <saza@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39316}
- Lowering the energy threshold for updating the accumulated error.
- Not using the pre-echo estimate in the initial frames when the matched filters have been recently initialized.
- Slight speed up for the increases in the accumulated error.
- Not periodically resetting the accumulated error.
Bug: webrtc:14205
Change-Id: Ic337332e263b27d7a3aba0ab4b371517780f9c90
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/291320
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Commit-Queue: Jesus de Vicente Pena <devicentepena@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39175}
Based on offline testing; needed to allow input volume adaptations
more frequently. Note that if the estimated speech level falls in
the target range, the recommended input volume won't change and
hence the new lower threshold won't necessarily increase the
number of adjustments.
Bug: webrtc:7494
Change-Id: Iabb501c188da238ea7b7137175bcfe09239c90a2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/291110
Reviewed-by: Hanna Silen <silen@webrtc.org>
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39161}
When the `WebRTC-Audio-GainController2` field trial is used, the
initial APM configuration is adjusted depending on its original
values and the field trial parameters.
This CL fixes two cases when the code crashes:
1. when, in the initial APM config, AGC1 is enabled, AGC2 is
disabled and TS is enabled
2. when the initial APM sample rate is different from the
capture one and the VAD APM sub-module is not re-initialized
This CL also improves the unit tests coverage and it has been
tested offline to check that the VAD sub-module is created only
when expected and that AGC2 uses its internal VAD when expected.
The tests ran on a few Wav files with different sample rates and
one AEC dump and on 16 different APM and field trial
configurations.
Bug: chromium:1407341, b/265112132
Change-Id: I7cc267ea81cb02be92c1f37f273b7ae93b6e4634
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/290988
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Reviewed-by: Olga Sharonova <olka@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39118}
- Test behavior with no input volume controller
- Test behavior with startup volume higher than the minimum
input volume
Bug: webrtc:7494
Change-Id: I36d48e2bd277b8a71eb6fbb0272c26c7176b3d5e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/286380
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Reviewed-by: Hanna Silen <silen@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38932}
Make sure that the input volume controller implementations exhibit
the adaptive behavior regardless of the sample rate and the number
of channels. The newly added tests check that:
- a downward adjustment takes place with clipping input
- an upward adjustment takes place with a too low speech level
- a downward adjustment takes place with a too high speech level
Bug: webrtc:14761
Change-Id: I1795e74c5f219e15107e928ebaca2bfa75214526
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/287301
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Reviewed-by: Hanna Silen <silen@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38930}
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}
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}
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}
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}
Now that `InputVolumeController` is finalized, it's time to
consolidate AGC2.
Main changes:
- Remove `AdaptiveDigitalGainController`: it's too simple to justify
a dedicated class and some components of it are also used by
`InputVolumeController`
- Remove unwanted temporal dependency: make `InputVolumeController`
adapt the volume based on the current speech level estimation and
not on the estimation from the previous frame
Tested: AGC2 adaptive digital bit-exactness verified
Bug: webrtc:7494
Change-Id: I175c2741cafc52be81794219c996a3824c3bbf5e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/280560
Reviewed-by: Hanna Silen <silen@webrtc.org>
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38841}
Regardless of the APM config, the transient suppressor (TS) submodule
won't be created if the `WebRTC-ApmTransientSuppressorKillSwitch`
field trial, disabled by default, is enabled.
Bug: webrtc:13663
Change-Id: Ic1ef9aa57c728296d671d4ef253630c581a86610
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/286382
Reviewed-by: Hanna Silen <silen@webrtc.org>
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38839}
Remove deprecated unit test helper functions CallPreProcessAudioBuffer()
and CallPreProcForChangingAudio(). Replace the use of these functions
with CallAgcSequence(). Remove a duplicate unit test using one of these
functions. The new calls follow the API contract.
Bug: webrtc:7494
Change-Id: Idc033cb48f4fab1814c4c6e0f23edc4a6a9faa64
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/285960
Reviewed-by: Alessio Bazzica <alessiob@webrtc.org>
Commit-Queue: Hanna Silen <silen@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38826}
Currently the fixed digital gain is applied after the input volume
controller and before the adaptive digital one. This CL moves its
application after the adaptive digital controller and before the
limiter.
Reasons:
- This change is safe: no production config where both adaptive and
fixed digital controllers are jointly used
- More predictable behavior: when the fixed digital controller is
used after the adaptive digital controller it is easier to describe
the overall behavior - i.e., the fixed digital combined with the
limiter can be used for digital compression
- Allow to remove an unwanted temporal dependency: in a follow-up CL
the input volume controller will use the latest speech level
estimation instead of that from the previously analyzed frame; this
CL makes that change easier.
Bug: webrtc:7494
Change-Id: I2e9869081e0eba1e4f30f11ea93a973ca7fea28c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/286340
Reviewed-by: Hanna Silen <silen@webrtc.org>
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38813}
Isolates the build targets for the `AdaptiveDigitalGainController`
dependencies that will be moved into `GainController2`.
`AdaptiveDigitalGainController` will be removed because the wrapper
itself adds little - that's the reason why it has no unit tests.
Bug: webrtc:7494
Change-Id: I2ca41f9255c8faefe4b2cb4ec1f8db536e582f39
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/280482
Reviewed-by: Hanna Silen <silen@webrtc.org>
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38799}
- use the new naming convention 'input volume'
- fix Yoda-style expressions in the unit tests
- clarify how the gain map values are generated
Bug: webrtc:7494
Change-Id: I4d6ee897a93cdefa6735733b053c57326d01a528
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/285467
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Reviewed-by: Hanna Silen <silen@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38795}
Add a field trial WebRTC-Audio-InputVolumeControllerExperiment and
a mechanism to adjust the config accordingly. Pass the additional
input volume controller config to GainController2.
Bug: webrtc:7494
Change-Id: I3dd624df1f4774cb533417747627995e1f60aa68
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/284101
Reviewed-by: Alessio Bazzica <alessiob@webrtc.org>
Reviewed-by: Per Åhgren <peah@webrtc.org>
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Commit-Queue: Hanna Silen <silen@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38780}
The integration relies on GainController2 methods Process() and
GetRecommendedInputVolume() to internally take into account whether
the input volume controller is enabled in the ctor or not. These
methods are called for every frame processed if GainController2 is
enabled. Analyze() is called if the input volume controller is
enabled.
The functionality can be enabled from the APM config and is not
enabled by default. If multiple input volume controllers are enabled,
an error is logged.
Tested: Bitexact on a large number of aecdumps if not enabled
Bug: webrtc:7494
Change-Id: I9105483be34eb95fab3c46afbbd368802e956fad
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/282720
Reviewed-by: Alessio Bazzica <alessiob@webrtc.org>
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Commit-Queue: Hanna Silen <silen@webrtc.org>
Reviewed-by: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38776}
Always enforce the minimum input volume, not only if overridden.
The only exception is when the applied input volume is zero: in that
case zero is still recommended.
This CL also adapts the unit tests and replaces "mic level" with
the "input volume".
Bug: webrtc:7494
Change-Id: I20c14624fbd357ab91ea05521c3723ec1045a8db
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/285462
Reviewed-by: Hanna Silen <silen@webrtc.org>
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38769}
Make the argument speech_probability non-optional in
InputVolumeController::Process() and
MonoInputVolumeController::Process().
Additional clean-up: Remove the flag enabled in the
config. Add unit tests for MonoInputVolumeController.
Bug: webrtc:7494
Change-Id: Ie28af77dc628bf71d09ce1ff033d39031f77a21e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/283700
Reviewed-by: Alessio Bazzica <alessiob@webrtc.org>
Commit-Queue: Hanna Silen <silen@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38710}
when volume emulation is used or when neither an input volume
controller nor volume emulation are used.
This CL adds 3 tests, 2 of which currently fail because APM
behaves in an undesired way. In [1] the behavior is fixed and
the tests are enabled.
A DCHECK in `AudioProcessingImpl::set_stream_analog_level` has
been removed since a more robust behavior can be obtained - namely,
that expected in the disabled unit tests added in this CL.
[1] https://webrtc-review.googlesource.com/c/src/+/281185
Bug: webrtc:14581
Change-Id: I29d2c000cd1baf90606487afd9a4042e6f487834
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/281184
Reviewed-by: Per Åhgren <peah@webrtc.org>
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38696}
Make speech probability threshold configurable by replacing
kSpeechProbabilitySilenceThreshold with speech_probability_threshold in
InputVolumeController::Config.
Make the processing more robust against outliers in speech probability
estimaton by computing an aggregate speech activity over a speech
segment. In MonoInputVolumeController::Process(), use the passed
non-empty speech probabilities to compute the speech activity over the
speech segment and only allow updates for segments with a high enough
ratio of speech frames. Pass RMS error and speech probability for every
frame in Process(): If rms_error_dbfs is empty, volume updates are not
allowed; if speech_probability is empty, the frame counts as a non-
speech frame.
Remove startup_min_volume from the config since it's no longer used
after https://webrtc-review.googlesource.com/c/src/+/282821.
Bug: webrtc:7494
Change-Id: I0ab81b03371496315348f552133aa9909bd36f26
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/283523
Commit-Queue: Hanna Silen <silen@webrtc.org>
Reviewed-by: Alessio Bazzica <alessiob@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38685}
This CL adds more explicit tests for unsupported sample rates in the WebRTC audio processing module (APM). Rates are restricted to the range [8000, 384000] Hz. Rates outside this range are handled as best as possible, depending on the format.
Tested: bitexact on a large number of aecdumps
Bug: chromium:1332484, chromium:1334991
Change-Id: I9639d03dc837e1fdff64d1f9d1fff0edc0fb299f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/276920
Commit-Queue: Sam Zackrisson <saza@webrtc.org>
Reviewed-by: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38663}
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}
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}
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}
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}
Implement InputVolumeController and RecommendedInputVolumeEstimator based on the copy of agc classes AgcManagerDirect and MonoAgc.
Copies of the original files created in https://webrtc-review.googlesource.com/c/src/+/278624.
Bug: webrtc:7494
Change-Id: I74acee57b0db5cc8a6b666be9ba619c6c98a1773
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/278625
Reviewed-by: Alessio Bazzica <alessiob@webrtc.org>
Commit-Queue: Hanna Silen <silen@webrtc.org>
Reviewed-by: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38533}
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 2235776597
$ 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 is a reland of commit 8d7273357d
Root cause:
audioproc_f doesn't call `metrics::Enable()` and therefore the stats
reporter crashed when `metrics::HistogramFactoryGetCountsLinear()`
returned a nullptr.
Bug fix:
Added `InputVolumeStatsReporter::cannot_log_stats_`, a const flag
that is set to true if any histogram factory returns a nullptr.
When true, the class does nothing.
This CL also includes other code readability improvements that were
not part of the original CL.
Original change's description:
> APM: log both applied and recommended input volume stats
>
> This CL replaces the existing `WebRTC.Audio.ApmAnalogGain.*` stats
> with `WebRTC.Audio.Apm.AppliedInputVolume.*` and adds the
> `WebRTC.Audio.Apm.RecommendedInputVolume.*` stats.
>
> Bug: webrtc:7494
> Change-Id: I70be710d20b1589fc814cbce3d3329ac1500686f
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/280220
> Reviewed-by: Hanna Silen <silen@webrtc.org>
> Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
> Cr-Commit-Position: refs/heads/main@{#38468}
Bug: webrtc:7494
Change-Id: I8373d16beb06b84f439d2c2274ededea7c5e95b0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/280661
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Reviewed-by: Hanna Silen <silen@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38484}
This CL replaces the existing `WebRTC.Audio.ApmAnalogGain.*` stats
with `WebRTC.Audio.Apm.AppliedInputVolume.*` and adds the
`WebRTC.Audio.Apm.RecommendedInputVolume.*` stats.
Bug: webrtc:7494
Change-Id: I70be710d20b1589fc814cbce3d3329ac1500686f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/280220
Reviewed-by: Hanna Silen <silen@webrtc.org>
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38468}
Adopt the new naming convention, which replaces "analog gain" and
"mic level" with "input volume", in the input volume stats reporter.
Bug: webrtc:7494
Change-Id: Ia24876151f51dd1dcc4e4f9db56c64d11ae3b442
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/279840
Reviewed-by: Hanna Silen <silen@webrtc.org>
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38467}
Enable by default the look for the first echo.
Bug: webrtc:14205
Change-Id: Iae904679c1432f3a0766263907cf376903685b97
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/278043
Reviewed-by: Jesus de Vicente Pena <devicentepena@webrtc.org>
Commit-Queue: Lionel Koenig <lionelk@webrtc.org>
Reviewed-by: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38452}
Tested: Chromium built with this change; verified that the
behavior at the beginning of the call has not changed with
both low (< 12) and high (> 12) input volumes.
Bug: webrtc:7494
Change-Id: Ie184c994d46bf6fd1cb209873383b911beb766e3
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/278787
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Reviewed-by: Jesus de Vicente Pena <devicentepena@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38420}
The parameter is unused and it will be removed in [1]. This CL
isolates the necessary unit test changes from [1].
[1] https://webrtc-review.googlesource.com/c/src/+/278787
Bug: webrtc:7494
Change-Id: Ic1179d335926fba8ff1b65b494b538cf849724bb
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/279100
Reviewed-by: Jesus de Vicente Pena <devicentepena@webrtc.org>
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38416}
This CL:
- makes it easier to understand the (nontrivial) metric interpretation
- corrects the computation of BufferDelay to use 0 for absent delay
- deletes metric MaxSkewShiftCount, unused since https://webrtc-review.googlesource.com/c/src/+/119701
- updates the unit test to directly test metric reporting
Corresponding update to histograms.xml:
https://crrev.com/c/3944909
Previous revert:
https://webrtc-review.googlesource.com/c/src/+/279040
This CL is identical to the original, except:
- the test is updated to spam fewer EXPECT_EQ failures on failure (EXPECT_EQs moved out of inner loop)
- the test not resets metrics (metrics::Reset()) at the beginning, like other histogram tests
Bug: webrtc:8671, chromium:1349051
Change-Id: Ie802e1f9d03a22ff7018f522a63b19e0b6eec2e8
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/279046
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Sam Zackrisson <saza@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38376}
- Set the initial input volume to that forced by startup min volume
since the latter is removed in a follow-up CL
- Remove unwanted expectations
Bug: webrtc:7494
Change-Id: I2df28f5bfaf4e592dfeae5e03b157268473cc822
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/278784
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Reviewed-by: Jesus de Vicente Pena <devicentepena@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38370}
This CL:
- makes it easier to understand the (nontrivial) metric interpretation
- corrects the computation of BufferDelay to use 0 for absent delay
- deletes metric MaxSkewShiftCount, unused since https://webrtc-review.googlesource.com/c/src/+/119701
- updates the unit test to directly test metric reporting
Corresponding update to histograms.xml:
https://crrev.com/c/3944909
Bug: webrtc:8671, chromium:1349051
Change-Id: If73b6fca4de7343bff2c53f72cedda458d36c599
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/278782
Commit-Queue: Sam Zackrisson <saza@webrtc.org>
Reviewed-by: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38362}
This CL adds #includes to header files in order to make them
self contained after the preprocessor pass.
Bug: b/251890128
Change-Id: I81c3ba38fb8ab8a2bbd151ba99aa871fae9f1b1b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/278422
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38327}
Remove the evaluation of clipping prediction. The result is not used.
Bug: webrtc:7494
Change-Id: I18d2c1f50ed675a9653d518095f69ed263a34041
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/278361
Reviewed-by: Alessio Bazzica <alessiob@webrtc.org>
Commit-Queue: Hanna Silen <silen@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38320}
Add passing optional speech level and speech probability to Process().
This enables computing an override for the RMS error from
Agc::GetRmsErrorDb(). Currently no speech level or probability are
passed outside the tests and no override happens elsewhere.
Bug: webrtc:7494
Change-Id: I0a7b1204aa51bcde8588963a5af023410405e83d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/277560
Reviewed-by: Alessio Bazzica <alessiob@webrtc.org>
Commit-Queue: Hanna Silen <silen@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38318}
Add a buffer class to store speech probabilities and to estimate speech
activity. Follows the implementation of speech activity computation in
LoudnessHistogram but uses floats for computations.
Bug: webrtc:7494
Change-Id: I6ee72ec52919904ea4e1fbe51d61993aa7813c9f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/277801
Reviewed-by: Alessio Bazzica <alessiob@webrtc.org>
Commit-Queue: Hanna Silen <silen@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38309}
MaybeInitializeCapture may overwrite the render configuration of a concurrent render reinitialization, leading to a second render reinitialization on the next render processing call.
See bug description for details.
Tested: Verified bitexactness offline (single-threaded) on a large number of aecdumps.
Bug: webrtc:14495
Change-Id: I9b70b454ce1c27859c3414c9c9ec89b7bbe35559
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/277380
Reviewed-by: Per Åhgren <peah@webrtc.org>
Commit-Queue: Sam Zackrisson <saza@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38241}
Main changes:
- `AgcManagerDirectTestHelper::FirstProcess()` replaced by
`CallAgcSequence()`, which is API contract compliant
- `ExpectCheckVolumeAndReset()`, `SetVolumeAndProcess()` and
`ExpectInitialize() `removed
- TODOs added for the next batch of improvements
- `AgcManagerDirectTestHelper::mock_agc` now using `NiceMock`
- `AgcManagerDirect::(AnalyzePre)Process()` now receives a
const ref
- `AnalyzePreProcess(const float* const*,size_t )` removed
Bug: webrtc:7494
Change-Id: Ie5bbaa590586dd806b30494fb00ca9c742c241e2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/273490
Reviewed-by: Hanna Silen <silen@webrtc.org>
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38056}
The current design of the modified getter is error-prone since the
returned value changes meaning based on when (which point in the code)
the getter is called - namely, before `ProcessStream()` is called the
getter returns the stream analog level, after it returns the
recommended level.
Plus, the new implementation, which essentially returns a local
member, removes the risks that the non-trivial implementation
is computationally expensive.
Bug: webrtc:7494, b/241923537
Change-Id: I6714444df27bcc055ae693974ecd1f77f79e6ec0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/271580
Reviewed-by: Hanna Silen <silen@webrtc.org>
Reviewed-by: Per Åhgren <peah@webrtc.org>
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38055}
The `recommended_stream_analog_level()` getter is used to retrieve
both the applied and the recommended input volume. This behavior is
error-prone since the caller must know what is returned based on
the point in the code (namely, before/after the AGC has changed
the last applied input volume into a recommended level).
This CL is a first step to make clarity on which input volume is
handled in different parts of APM. Next in the pipeline: make
`recommended_stream_analog_level()` a trivial getter that always
returns the recommended level.
Main changes:
- When `recommended_stream_analog_level()` is called but
`set_stream_analog_level()` is not called, APM logs an error
and returns a fall-back volume (which should not be applied
since, when `set_stream_analog_level()` is not called, no
external input volume is expected to be present
- When APM is used without calling the `*_stream_analog_level()`
methods (e.g., when the caller does not provide any input volume),
the recorded AEC dumps won't store `Stream::applied_input_level`
Other changes:
- Removed `AudioProcessingImpl::capture_::prev_analog_mic_level`
- Removed redundant code in `GainController2` around detecting
input volume changes (already done by APM)
- Adapted the `audioproc_f` and `unpack_aecdump` tools
- Data dumps clean-up: the applied and the recommended input
volumes are now recorded in an AGC implementation agnostic way
Bug: webrtc:7494, b/241923537
Change-Id: I3cb4a731fd9f3dc19bf6ac679b7ed8c969ea283b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/271544
Reviewed-by: Per Åhgren <peah@webrtc.org>
Reviewed-by: Hanna Silen <silen@webrtc.org>
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38054}
Switching to an AGC implementation agnostic solution for the input
volume emulation functionality offered by the
`capture_levels_adjuster` sub-module.
This CL also fixes a (silent) bug due to which, when the input
volume is emulated via the capture adjuster sub-module, AGC2
reads an incorrect value for the applied input volume.
Tested: audioproc_f with `--analog_mic_gain_emulation 1` used
to verify bit-exactness for one Wav file and one AEC dump for
which the input volume varies.
Bug: webrtc:7494, b/241923537
Change-Id: Ide3085f9a5dfd85888ad812ebd56faa175fb2ba7
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/273902
Reviewed-by: Per Åhgren <peah@webrtc.org>
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38053}
`EchoControllerObservesAnalogAgc1EchoPathGainChange` is incorrect
since it does not call `set_stream_analog_level()`,
`ProcessCapture()` and `recommended_stream_analog_level()`
according to the contract.
`EchoControllerObservesNoDigitalAgc2EchoPathGainChange` is
useless since AGC2 doesn't have any analog controller at the
moment and the test is not written to explictly trigger digital
gain adaptations.
Bug: webrtc:7494, b/241923537
Change-Id: I56203c736448ec060077b00b57e98cd4c29fa737
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/271541
Reviewed-by: Hanna Silen <silen@webrtc.org>
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38052}
Making it clear that the field is used to store the applied input
volume and not the recommended input volume.
Bug: webrtc:7494, b/241923537
Change-Id: Ib91bc1a12348f63e3a4ba6e068ed02e40786a87b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/271342
Reviewed-by: Jesus de Vicente Pena <devicentepena@webrtc.org>
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38051}
Fixing a bug due to which the analog controller could not be disabled.
AudioProcessing::Config::GainController1::AnalogGainController::enabled
was ignored and therefore `recommended_stream_analog_level_locked()` in
APM was returning the level recommended by `AgcManagerDirect`.
When the analog controller is disabled, `stream_analog_level()` now
returns the last value set via `set_stream_analog_level()`.
However, the analog controller code is still running and, in particular,
the existing metrics are reported as if the controller were enabled.
This choice was made to reduce the risks of adding bugs in the digital
compression gain selection part, which is tied to the analog
controller. The metric drawback will be solved in a follow-up CL.
Additional changes:
- log `WebRTC.Audio.GainController.Analog.Enabled` when
AGC1 is created or when its config changes
- first step to replace "analog level" with "input volume"
Bug: webrtc:7909, b/180019868
Change-Id: I28ce9556dd98f3dd9ad546799406c55478730435
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/270663
Reviewed-by: Per Åhgren <peah@webrtc.org>
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38044}
