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}
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}
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}
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}
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}
Add a VoiceActivityDetectorWrapper submodule in AudioProcessingImpl
and enable injecting speech probability into GainController2.
Bug: webrtc:13663
Change-Id: I05e13b737d085b45ac8ce76660191867c56834c2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/265166
Commit-Queue: Hanna Silen <silen@webrtc.org>
Reviewed-by: Alessio Bazzica <alessiob@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#37275}
The class has also been renamed to better reflect its purpose.
Bug: webrtc:7494
Change-Id: I223a364ab4f8b8a5fef765848bf05675d045cefd
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/236343
Reviewed-by: Hanna Silen <silen@webrtc.org>
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35277}
Bit exactness verified with audioproc_f on a collection of AEC dumps
and Wav files (42 recordings in total).
Bug: webrtc:7494
Change-Id: Id9849c4463791f5a203afe31efc163efb4d4458e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/234583
Reviewed-by: Hanna Silen <silen@webrtc.org>
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35248}
When `AudioProcessingImpl::ApplyConfig()` is called, AGC2 is initialized
and then the new config is applied. That is error prone and for example
breaks bit exactness in [1].
Changes:
- `GainController2` must be created by passing configuration,
sample rate and number of channels
- `GainController2::ApplyConfig()` removed
Bit exactness verified with audioproc_f on a collection of AEC dumps
and Wav files (42 recordings in total).
[1] https://webrtc-review.googlesource.com/c/src/+/234587.
Bug: webrtc:7494
Change-Id: I251e03603394a4fc8769b9b5c197a157893676a9
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/235060
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Reviewed-by: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35206}
Add the option to run the adaptive digital controller of AGC2 without
side-effects - i.e., no gain applied.
Tested: adapation verified during a video call in chromium
Bug: webrtc:7494
Change-Id: I4776f6012907d76a17a3bca89991da97dc38657f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/215964
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Reviewed-by: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33875}
The APM config to string mapping must be in one place (namely,
in `audio_processing.cc`). This CL moves the AGC2 config to string
impl to the right place.
This CL also updates `GainController2::Validate()` and adds the
missing unit tests for the parameters that have recently been added.
Stack buffer size in `AudioProcessing::Config::ToString()` increased
because of the extra params. Syntax near `multi_channel_capture` fixed.
Output string format verified with a JS linter.
Bug: webrtc:7494
Change-Id: I692e1549b7d40c970d88a14c8e83da16325fb54c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/187080
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Reviewed-by: Minyue Li <minyue@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32400}
This CL refactors AGC2 and fixes the order with which the fixed
and the adaptive digital gain controllers are applied - i.e., fixed
first, then adaptive and finally limiter.
FixedGainController has been removed since we need to split the
processing done by the gain applier and the limiter.
Also, GainApplier and Limiter are easy enough to be used without
a wrapper and a wrapper would need 2 separated calls in the right
order - i.e., error prone.
FrameCombiner in audio mixer has been adapted and now only uses the
limiter (which is what is needed since no gain is applied).
The unit tests for FixedGainController have been moved to
gain_controller2_unittests. They have been re-adapted and
ChangeFixedGainShouldBeFastAndTimeInvariant has been re-tuned.
Bug: webrtc:7494
Change-Id: I4d7daeae917257ac019a645b74deba6642f77322
Reviewed-on: https://webrtc-review.googlesource.com/c/108624
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Reviewed-by: Alex Loiko <aleloi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25477}
The extra saturation margin is a setting for the SaturationProtector
in GainController2. The higher it is, the less gain GC2 will apply. In
this CL we pipe the setting up to audio_processing.h. Now the setting
can be set at a high level.
Also in this CL add a few (missing, they should have been there
already) tests for the GC2 and GC2 with saturation margin.
Bug: webrtc:7494
Change-Id: I1b61f1662e6c6a8817fd5b0e845339694bf8d50d
Reviewed-on: https://webrtc-review.googlesource.com/c/109001
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Commit-Queue: Alex Loiko <aleloi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25470}
The AGC2 is enabled by flipping
AudioProcessing::Config::GainController2::enabled. The flag enables
both AdaptiveAgc and FixedGainController. Before this CL, there was no
way(*) to only enable the FixedGainController. After this CL, it's
also possible to flip the setting
|AudioProcessing::Config::GainController2::adaptive_digital_mode|. The
default is |true|, which is the previous behavior.
* Except for instantiating and setting it up outside of the APM like
it's done in the AudioMixer.
Bug: webrtc:7494
Change-Id: I506e93b6687221ac467f083fa8db3d45c98c1b83
Reviewed-on: https://webrtc-review.googlesource.com/95426
Commit-Queue: Alex Loiko <aleloi@webrtc.org>
Reviewed-by: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24432}
Agc2 applies a digital gain to the nearend signal.
When the analog level changes, the digital gain calculation is no
longer valid. Therefore Agc2 should be notified to analog gain
changes.
This CL also allow audioproc_f to chain AGC1 and AGC2. In a dependent
CL we will allow using AGC1 for analog gain and AGC2 for digital
gain.
Bug: webrtc:7494
Change-Id: Id75b3728fbf2de1d84b7fba005e4670c7a2985d9
Reviewed-on: https://webrtc-review.googlesource.com/89387
Commit-Queue: Alex Loiko <aleloi@webrtc.org>
Reviewed-by: Alessio Bazzica <alessiob@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24231}
We update the configuration settings for AGC2. We also update their
effects. Now, 'gain_controller2.enable=true' means 'first run Adaptive
AGC2; then run AGC2 limiter'.
Previously, only the AGC2 limiter was implemented. To run that, one
had to set both 'gain_controller2.enable=true' and
'gain_controller2.enable_limiter=true'.
This setting also enables adaptive AGC2 in the test tool 'audioproc_f'.
Bug: webrtc:7494
Change-Id: I0d5dfe443f2cdc0ecf3aa4054442dab6276d284d
Reviewed-on: https://webrtc-review.googlesource.com/64990
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Commit-Queue: Alex Loiko <aleloi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22669}
The FixedGainController (FGC) applies a fixed gain. It will also
control the limiter. The limiter will be landed over the next several
CLs.
The GainController2 is a 'private submodule' of APM. It will control
the new automatic gain controller (AGC). It controls the AGC through
Initialize() and ApplyConfig().
This CL contains
* build changes to make modules/audio_processing/agc2 an independent
target
* a new MutableFloatAudioFrame which is the audio interface between
AGC2 and APM
* move of the fixed gain application from GainController2 to
FixedGainController.
If you are a googler, there is more information in this doc:
https://docs.google.com/document/d/1RV2Doet3MZtUPAHVva61Vjo20iyd1bmmm3aR8znWpzo/edit#
Bug: webrtc:7949
Change-Id: Ief95cbbce83c3aafe54638fd2ab881c9fb8bdc3a
Reviewed-on: https://webrtc-review.googlesource.com/50440
Commit-Queue: Alex Loiko <aleloi@webrtc.org>
Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22046}
2018-02-16 10:56:38 +00:00
Renamed from modules/audio_processing/agc2/gain_controller2.h (Browse further)
