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}
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 is a reland of commit 8d7273357d92fab881561d886ce8dfe94e6e2238
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}
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}
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}
