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}
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}
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}
- 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}
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}
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}
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 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}
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}
