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webrtc_m130/modules/audio_processing/agc2/BUILD.gn

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Add FixedGainController and move GainController2 in APM. 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 11:54:07 +01:00
# Copyright (c) 2017 The WebRTC project authors. All Rights Reserved.
#
# Use of this source code is governed by a BSD-style license
# that can be found in the LICENSE file in the root of the source
# tree. An additional intellectual property rights grant can be found
# in the file PATENTS. All contributing project authors may
# be found in the AUTHORS file in the root of the source tree.
import("../../../webrtc.gni")
Adaptive Digital gain control structure. This CL defines the control flow of the adaptive AGC. It also defines method and class stubs. Contents: 1. Divide the 'agc2' build target into 'fixed_digital' and 'adaptive_digital'. 1. Update the dependencies of everything that depended on 'agc2'. 2. Define the sub-modules of the adaptive digital AGC 2. They are: 1. Level Estimator - it gets the energy and a speech probability and updates a speech level estimate. 2. Noise Estimator - it gets an immutable view of the speech frame and updates the noise level estimate 3. Gain applier - it gets the speech frame, the current speech and noise estimates, and the speech probability. It finds a gain to apply and applies it. 4. AdaptiveAgc - sets up and controls the sub-modules described above. Bug: webrtc:7494 Change-Id: Ib7ccd8924e94eead0bc5f935b5d8a12e06e24fd1 Reviewed-on: https://webrtc-review.googlesource.com/64440 Reviewed-by: Alessio Bazzica <alessiob@webrtc.org> Commit-Queue: Alex Loiko <aleloi@webrtc.org> Cr-Commit-Position: refs/heads/master@{#22628}
2018-03-27 13:38:36 +02:00
group("agc2") {
deps = [
":adaptive_digital",
":fixed_digital",
]
}
Allow AGC2 level estimation in AgcManagerDirect. This CL does the following: 1. Adds a new AdaptiveModeLevelEstimatorAgc implementation of the Agc interface. The new implementation differs from webrtc::Agc by 1. using the AGC2 speech level estimator in GetRmsErrorDb. webrtc::Agc implements its own with help of webrtc::LoudnessHistogram. 2. Doesn't forget its past at every GetRmsErrorDb call. 2. Makes AgcManagerDirect use AdaptiveModeLevelEstimatorAgc instead of webrtc::Agc if the use_agc2_level_estimation flag is set. Bug: webrtc:7494 Change-Id: I8df3f52e322d433eb5ce5297f4236af2f1877b04 Reviewed-on: https://webrtc-review.googlesource.com/86603 Commit-Queue: Alex Loiko <aleloi@webrtc.org> Reviewed-by: Sam Zackrisson <saza@webrtc.org> Cr-Commit-Position: refs/heads/master@{#23875}
2018-07-06 15:35:42 +02:00
rtc_source_set("level_estimation_agc") {
sources = [
"adaptive_mode_level_estimator_agc.cc",
"adaptive_mode_level_estimator_agc.h",
]
configs += [ "..:apm_debug_dump" ]
deps = [
":adaptive_digital",
":common",
":gain_applier",
":noise_level_estimator",
":rnn_vad_with_level",
"..:apm_logging",
"..:audio_frame_view",
"../../../api:array_view",
"../../../common_audio",
"../../../rtc_base:checks",
"../../../rtc_base:rtc_base_approved",
"../../../rtc_base:safe_minmax",
"../agc:level_estimation",
"../vad",
]
}
Adaptive Digital gain control structure. This CL defines the control flow of the adaptive AGC. It also defines method and class stubs. Contents: 1. Divide the 'agc2' build target into 'fixed_digital' and 'adaptive_digital'. 1. Update the dependencies of everything that depended on 'agc2'. 2. Define the sub-modules of the adaptive digital AGC 2. They are: 1. Level Estimator - it gets the energy and a speech probability and updates a speech level estimate. 2. Noise Estimator - it gets an immutable view of the speech frame and updates the noise level estimate 3. Gain applier - it gets the speech frame, the current speech and noise estimates, and the speech probability. It finds a gain to apply and applies it. 4. AdaptiveAgc - sets up and controls the sub-modules described above. Bug: webrtc:7494 Change-Id: Ib7ccd8924e94eead0bc5f935b5d8a12e06e24fd1 Reviewed-on: https://webrtc-review.googlesource.com/64440 Reviewed-by: Alessio Bazzica <alessiob@webrtc.org> Commit-Queue: Alex Loiko <aleloi@webrtc.org> Cr-Commit-Position: refs/heads/master@{#22628}
2018-03-27 13:38:36 +02:00
rtc_source_set("adaptive_digital") {
sources = [
"adaptive_agc.cc",
"adaptive_agc.h",
"adaptive_digital_gain_applier.cc",
"adaptive_digital_gain_applier.h",
"adaptive_mode_level_estimator.cc",
"adaptive_mode_level_estimator.h",
Level estimation and saturation protection stub. The level estimator (AdaptiveModeLevelEstimator) produces a biased estimate of the speech level. In our model, we use another module (the SaturationProtector) to compute the bias. This CL contains the estimator and a stub of the saturation protector. Bug: webrtc:7494 Change-Id: I0df736d0346063f544fa680b4cc84177ea548545 Reviewed-on: https://webrtc-review.googlesource.com/64820 Commit-Queue: Alex Loiko <aleloi@webrtc.org> Reviewed-by: Ivo Creusen <ivoc@webrtc.org> Cr-Commit-Position: refs/heads/master@{#22641}
2018-03-28 09:45:29 +02:00
"saturation_protector.cc",
"saturation_protector.h",
Adaptive Digital gain control structure. This CL defines the control flow of the adaptive AGC. It also defines method and class stubs. Contents: 1. Divide the 'agc2' build target into 'fixed_digital' and 'adaptive_digital'. 1. Update the dependencies of everything that depended on 'agc2'. 2. Define the sub-modules of the adaptive digital AGC 2. They are: 1. Level Estimator - it gets the energy and a speech probability and updates a speech level estimate. 2. Noise Estimator - it gets an immutable view of the speech frame and updates the noise level estimate 3. Gain applier - it gets the speech frame, the current speech and noise estimates, and the speech probability. It finds a gain to apply and applies it. 4. AdaptiveAgc - sets up and controls the sub-modules described above. Bug: webrtc:7494 Change-Id: Ib7ccd8924e94eead0bc5f935b5d8a12e06e24fd1 Reviewed-on: https://webrtc-review.googlesource.com/64440 Reviewed-by: Alessio Bazzica <alessiob@webrtc.org> Commit-Queue: Alex Loiko <aleloi@webrtc.org> Cr-Commit-Position: refs/heads/master@{#22628}
2018-03-27 13:38:36 +02:00
]
configs += [ "..:apm_debug_dump" ]
deps = [
":common",
Pre-amplification in the audio processing module. Added a new sub-module 'GainApplier'. The build target is 'modules/audio_processing/agc2:gain_applier'. A small refactoring makes the GainApplier used in adaptive-digital AGC2. The AGC2 now multiplies samples with a gain in 3 places. It's the GainApplier, the GainCurveApplier, and the FixedGainController. The GainApplier is used in AdaptiveDigitalGainApplier and will be used as a pre-amplifier. Bug: webrtc:9138 Change-Id: Ibc4c0ea109c6757f159d4adb6e3d8614179c9bc6 Reviewed-on: https://webrtc-review.googlesource.com/69321 Commit-Queue: Alex Loiko <aleloi@webrtc.org> Reviewed-by: Alessio Bazzica <alessiob@webrtc.org> Cr-Commit-Position: refs/heads/master@{#22849}
2018-04-13 11:15:34 +02:00
":gain_applier",
Noise level estimation for AGC2. We put back the old noise estimator from LevelController. We add a few new unit tests. We also re-arrange the code so that it fits with how it is used in AGC2. The differences are: 1. The NoiseLevelEstimator is now fully self-contained. 2. The NoiseLevelEstimator is responsible for calling SignalClassifier and computing the signal energy. Previously the signal type and energy were used in several places. It made sense to compute the values independently of the noise calculation. 3. Re-initialization doesn't have to be done by the caller. 4. The interface is AudioFrameView instead of AudioBuffer. # Bots are green, nothing should break internal stuff NOTRY=True Bug: webrtc:7494 Change-Id: I442bdbbeb3796eb2518e96000aec9dc5a039ae6d Reviewed-on: https://webrtc-review.googlesource.com/66380 Commit-Queue: Alex Loiko <aleloi@webrtc.org> Reviewed-by: Sam Zackrisson <saza@webrtc.org> Cr-Commit-Position: refs/heads/master@{#22738}
2018-04-04 15:05:57 +02:00
":noise_level_estimator",
Add RNN-VAD to AGC2. * Move 'VadWithLevel' to AGC2 where it belongs. * Remove the vectors from VadWithLevel. They were there to make it work with modules/audio_processing/vad, which we don't need any longer. * Remove the vector handling from AGC2. It was spread out across AdaptiveDigitalGainApplier, AdaptiveAGC and their unit tests. * Hack the RNN VAD into VadWithLevel. The main issue is the resampling. Bug: webrtc:9076 Change-Id: I13056c985d0ec41269735150caf4aaeb6ff9281e Reviewed-on: https://webrtc-review.googlesource.com/77364 Reviewed-by: Sam Zackrisson <saza@webrtc.org> Commit-Queue: Alex Loiko <aleloi@webrtc.org> Cr-Commit-Position: refs/heads/master@{#23688}
2018-06-20 14:14:18 +02:00
":rnn_vad_with_level",
Adaptive Digital gain control structure. This CL defines the control flow of the adaptive AGC. It also defines method and class stubs. Contents: 1. Divide the 'agc2' build target into 'fixed_digital' and 'adaptive_digital'. 1. Update the dependencies of everything that depended on 'agc2'. 2. Define the sub-modules of the adaptive digital AGC 2. They are: 1. Level Estimator - it gets the energy and a speech probability and updates a speech level estimate. 2. Noise Estimator - it gets an immutable view of the speech frame and updates the noise level estimate 3. Gain applier - it gets the speech frame, the current speech and noise estimates, and the speech probability. It finds a gain to apply and applies it. 4. AdaptiveAgc - sets up and controls the sub-modules described above. Bug: webrtc:7494 Change-Id: Ib7ccd8924e94eead0bc5f935b5d8a12e06e24fd1 Reviewed-on: https://webrtc-review.googlesource.com/64440 Reviewed-by: Alessio Bazzica <alessiob@webrtc.org> Commit-Queue: Alex Loiko <aleloi@webrtc.org> Cr-Commit-Position: refs/heads/master@{#22628}
2018-03-27 13:38:36 +02:00
"..:apm_logging",
"..:audio_frame_view",
"../../../api:array_view",
"../../../common_audio",
"../../../rtc_base:checks",
"../../../rtc_base:rtc_base_approved",
"../../../rtc_base:safe_minmax",
Gain metrics for digital adaptive AGC. We add 2 metrics for measuring applied digital gain to AgcManagerDirect. We also add an applied gain and an estimated noise metric to Agc2. Chromium histogram CL is https://chromium-review.googlesource.com/c/chromium/src/+/1170833 Bug: webrtc:7494 Change-Id: Ie40873f9e43bc7d34d8f5473cd73bd47eb84e855 Reviewed-on: https://webrtc-review.googlesource.com/93468 Reviewed-by: Ivo Creusen <ivoc@webrtc.org> Commit-Queue: Alex Loiko <aleloi@webrtc.org> Cr-Commit-Position: refs/heads/master@{#24290}
2018-08-10 14:43:51 +02:00
"../../../system_wrappers:metrics_api",
Adaptive Digital gain control structure. This CL defines the control flow of the adaptive AGC. It also defines method and class stubs. Contents: 1. Divide the 'agc2' build target into 'fixed_digital' and 'adaptive_digital'. 1. Update the dependencies of everything that depended on 'agc2'. 2. Define the sub-modules of the adaptive digital AGC 2. They are: 1. Level Estimator - it gets the energy and a speech probability and updates a speech level estimate. 2. Noise Estimator - it gets an immutable view of the speech frame and updates the noise level estimate 3. Gain applier - it gets the speech frame, the current speech and noise estimates, and the speech probability. It finds a gain to apply and applies it. 4. AdaptiveAgc - sets up and controls the sub-modules described above. Bug: webrtc:7494 Change-Id: Ib7ccd8924e94eead0bc5f935b5d8a12e06e24fd1 Reviewed-on: https://webrtc-review.googlesource.com/64440 Reviewed-by: Alessio Bazzica <alessiob@webrtc.org> Commit-Queue: Alex Loiko <aleloi@webrtc.org> Cr-Commit-Position: refs/heads/master@{#22628}
2018-03-27 13:38:36 +02:00
]
}
AGC2 Bi-Quad filter: separate target and unit test. Adding a build target for the bi-qaud filter to make it available for the RNN VAD of AGC2. Also adding a unit test to test the computation both in-place and not in-place while comparing the produced output to that of scipy.signal. Bug: webrtc:9076 Change-Id: I16176a477ee4b81bb1e090c4906c3a9948ad2772 Reviewed-on: https://webrtc-review.googlesource.com/74220 Commit-Queue: Alessio Bazzica <alessiob@webrtc.org> Reviewed-by: Alex Loiko <aleloi@webrtc.org> Cr-Commit-Position: refs/heads/master@{#23141}
2018-05-07 13:26:47 +02:00
rtc_source_set("biquad_filter") {
AGC2 RNN VAD: Feature extraction. This CL finalizes the feature extraction part for the RNN VAD adding a class that combines a high-pass filter, LP residual computation, pitch estimation and spectral features extraction. This CL also includes a minor refactoring of the pitch estimation library. Bug: webrtc:9076 Change-Id: I918b9e143bc6dd2bf508a891446067258a68a777 Reviewed-on: https://webrtc-review.googlesource.com/75504 Commit-Queue: Alessio Bazzica <alessiob@webrtc.org> Reviewed-by: Alex Loiko <aleloi@webrtc.org> Cr-Commit-Position: refs/heads/master@{#23235}
2018-05-14 20:22:18 +02:00
visibility = [ "./*" ]
AGC2 Bi-Quad filter: separate target and unit test. Adding a build target for the bi-qaud filter to make it available for the RNN VAD of AGC2. Also adding a unit test to test the computation both in-place and not in-place while comparing the produced output to that of scipy.signal. Bug: webrtc:9076 Change-Id: I16176a477ee4b81bb1e090c4906c3a9948ad2772 Reviewed-on: https://webrtc-review.googlesource.com/74220 Commit-Queue: Alessio Bazzica <alessiob@webrtc.org> Reviewed-by: Alex Loiko <aleloi@webrtc.org> Cr-Commit-Position: refs/heads/master@{#23141}
2018-05-07 13:26:47 +02:00
sources = [
"biquad_filter.cc",
"biquad_filter.h",
]
deps = [
"../../../api:array_view",
"../../../rtc_base:rtc_base_approved",
]
}
rtc_source_set("common") {
sources = [
"agc2_common.h",
]
deps = [
"../../../rtc_base:rtc_base_approved",
]
}
Adaptive Digital gain control structure. This CL defines the control flow of the adaptive AGC. It also defines method and class stubs. Contents: 1. Divide the 'agc2' build target into 'fixed_digital' and 'adaptive_digital'. 1. Update the dependencies of everything that depended on 'agc2'. 2. Define the sub-modules of the adaptive digital AGC 2. They are: 1. Level Estimator - it gets the energy and a speech probability and updates a speech level estimate. 2. Noise Estimator - it gets an immutable view of the speech frame and updates the noise level estimate 3. Gain applier - it gets the speech frame, the current speech and noise estimates, and the speech probability. It finds a gain to apply and applies it. 4. AdaptiveAgc - sets up and controls the sub-modules described above. Bug: webrtc:7494 Change-Id: Ib7ccd8924e94eead0bc5f935b5d8a12e06e24fd1 Reviewed-on: https://webrtc-review.googlesource.com/64440 Reviewed-by: Alessio Bazzica <alessiob@webrtc.org> Commit-Queue: Alex Loiko <aleloi@webrtc.org> Cr-Commit-Position: refs/heads/master@{#22628}
2018-03-27 13:38:36 +02:00
rtc_source_set("fixed_digital") {
Add FixedGainController and move GainController2 in APM. 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 11:54:07 +01:00
sources = [
AGC2-fixed-digital: Level Estimator This CL adds the Level Estimator of the new gain controller. The Level Estimator divides a 10ms input frame in kSubFramesInFrame=20 sub frames. We take the maximal sample values in every sub frame. We then apply attack/decay smoothing. This is the final level estimate. The results will be used with InterpolatedGainCurve (see this CL https://webrtc-review.googlesource.com/c/src/+/51920). For every level estimate value, we look up a gain with InterpolatedGainCurve::LookUpGainToApply. This gain is then applied to the signal. Bug: webrtc:7949 Change-Id: I2b4b3894a3e945d3dd916ce516c79abacb2b18b1 Reviewed-on: https://webrtc-review.googlesource.com/52381 Reviewed-by: Sam Zackrisson <saza@webrtc.org> Commit-Queue: Alex Loiko <aleloi@webrtc.org> Cr-Commit-Position: refs/heads/master@{#22054}
2018-02-16 12:39:00 +01:00
"fixed_digital_level_estimator.cc",
"fixed_digital_level_estimator.h",
Add FixedGainController and move GainController2 in APM. 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 11:54:07 +01:00
"fixed_gain_controller.cc",
"fixed_gain_controller.h",
Fixed Digital mode of AGC2 implementation finished. This CL adds the GainCurveApplier (GCA). It owns a FixedDigitalLevelEstimator (LE) and an InterpolatedGainCurve (IGC). The GCA uses the LE to compute the input signal level, looks up a gain from IGC and applies it on the signal. The other IGC and LE submodules were added in previous CLs [1] and [2]. This CL also turns on AGC2 in the APM fuzzer. [1] https://webrtc-review.googlesource.com/c/src/+/51920 [2] https://webrtc-review.googlesource.com/c/src/+/52381 Bug: webrtc:7949 Change-Id: Idb10cc3ca9d6d2e4ac5824cc3391ed8aa680f6cd Reviewed-on: https://webrtc-review.googlesource.com/54361 Commit-Queue: Alex Loiko <aleloi@webrtc.org> Reviewed-by: Sam Zackrisson <saza@webrtc.org> Cr-Commit-Position: refs/heads/master@{#22103}
2018-02-20 15:58:36 +01:00
"gain_curve_applier.cc",
"gain_curve_applier.h",
"interpolated_gain_curve.cc",
"interpolated_gain_curve.h",
Add FixedGainController and move GainController2 in APM. 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 11:54:07 +01:00
]
configs += [ "..:apm_debug_dump" ]
deps = [
Adaptive Digital gain control structure. This CL defines the control flow of the adaptive AGC. It also defines method and class stubs. Contents: 1. Divide the 'agc2' build target into 'fixed_digital' and 'adaptive_digital'. 1. Update the dependencies of everything that depended on 'agc2'. 2. Define the sub-modules of the adaptive digital AGC 2. They are: 1. Level Estimator - it gets the energy and a speech probability and updates a speech level estimate. 2. Noise Estimator - it gets an immutable view of the speech frame and updates the noise level estimate 3. Gain applier - it gets the speech frame, the current speech and noise estimates, and the speech probability. It finds a gain to apply and applies it. 4. AdaptiveAgc - sets up and controls the sub-modules described above. Bug: webrtc:7494 Change-Id: Ib7ccd8924e94eead0bc5f935b5d8a12e06e24fd1 Reviewed-on: https://webrtc-review.googlesource.com/64440 Reviewed-by: Alessio Bazzica <alessiob@webrtc.org> Commit-Queue: Alex Loiko <aleloi@webrtc.org> Cr-Commit-Position: refs/heads/master@{#22628}
2018-03-27 13:38:36 +02:00
":common",
Add FixedGainController and move GainController2 in APM. 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 11:54:07 +01:00
"..:apm_logging",
"..:audio_frame_view",
"../../../api:array_view",
"../../../common_audio",
"../../../rtc_base:checks",
Fixed Digital mode of AGC2 implementation finished. This CL adds the GainCurveApplier (GCA). It owns a FixedDigitalLevelEstimator (LE) and an InterpolatedGainCurve (IGC). The GCA uses the LE to compute the input signal level, looks up a gain from IGC and applies it on the signal. The other IGC and LE submodules were added in previous CLs [1] and [2]. This CL also turns on AGC2 in the APM fuzzer. [1] https://webrtc-review.googlesource.com/c/src/+/51920 [2] https://webrtc-review.googlesource.com/c/src/+/52381 Bug: webrtc:7949 Change-Id: Idb10cc3ca9d6d2e4ac5824cc3391ed8aa680f6cd Reviewed-on: https://webrtc-review.googlesource.com/54361 Commit-Queue: Alex Loiko <aleloi@webrtc.org> Reviewed-by: Sam Zackrisson <saza@webrtc.org> Cr-Commit-Position: refs/heads/master@{#22103}
2018-02-20 15:58:36 +01:00
"../../../rtc_base:gtest_prod",
Add FixedGainController and move GainController2 in APM. 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 11:54:07 +01:00
"../../../rtc_base:rtc_base_approved",
Separate build target for rtc_base/numerics/safe_minmax.h So that we can avoid dependency cycles. Bug: none Change-Id: I821d9f1319dff01403d6e4e310cbb2d4b2b125e8 Reviewed-on: https://webrtc-review.googlesource.com/60500 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#22328}
2018-03-07 14:18:56 +01:00
"../../../rtc_base:safe_minmax",
Add more Audio Mixer and Fixed Gain Controller metrics. We want to know how the AudioMixer is used and how FixedGainController behaves. The WebRTC.Audio.Agc2.FixedDigitalGainCurveRegion.* metrics measures how often the input level hits different regions of the Fixed Gain Controller gain curve (when the limiter is enabled). They also measure how long the metrics stay in different regions. They are related to WebRTC.Audio.ApmCaptureOutputLevelPeakRms, but the new metrics measure the level before any processing done in APM. The AudioMixer mixes incoming audio streams. Their number should be mostly constant, and often some of them could be muted. The metrics WebRTC.Audio.AudioMixer.NumIncomingStreams, WebRTC.Audio.AudioMixer.NumIncomingActiveStreams log the number of incoming stream and how many are not muted. We currently don't have any stats related to that. The metric WebRTC.Audio.AudioMixer.MixingRate logs the rate selected for mixing. The rate can sometimes be inferred from WebRTC.Audio.Encoder.CodecType. But that metric measures encoding and not decoding, and codecs don't always map to rates. See also accompanying Chromium CL https://chromium-review.googlesource.com/c/chromium/src/+/939473 Bug: webrtc:8925 Change-Id: Ib1405877fc1b39e5d2f0ceccba04434813f20b0d Reviewed-on: https://webrtc-review.googlesource.com/57740 Reviewed-by: Alessio Bazzica <alessiob@webrtc.org> Commit-Queue: Alex Loiko <aleloi@webrtc.org> Cr-Commit-Position: refs/heads/master@{#22443}
2018-03-14 12:27:05 +01:00
"../../../system_wrappers:metrics_api",
Add FixedGainController and move GainController2 in APM. 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 11:54:07 +01:00
]
}
AGC2 Bi-Quad filter: separate target and unit test. Adding a build target for the bi-qaud filter to make it available for the RNN VAD of AGC2. Also adding a unit test to test the computation both in-place and not in-place while comparing the produced output to that of scipy.signal. Bug: webrtc:9076 Change-Id: I16176a477ee4b81bb1e090c4906c3a9948ad2772 Reviewed-on: https://webrtc-review.googlesource.com/74220 Commit-Queue: Alessio Bazzica <alessiob@webrtc.org> Reviewed-by: Alex Loiko <aleloi@webrtc.org> Cr-Commit-Position: refs/heads/master@{#23141}
2018-05-07 13:26:47 +02:00
rtc_source_set("gain_applier") {
Adaptive Digital gain control structure. This CL defines the control flow of the adaptive AGC. It also defines method and class stubs. Contents: 1. Divide the 'agc2' build target into 'fixed_digital' and 'adaptive_digital'. 1. Update the dependencies of everything that depended on 'agc2'. 2. Define the sub-modules of the adaptive digital AGC 2. They are: 1. Level Estimator - it gets the energy and a speech probability and updates a speech level estimate. 2. Noise Estimator - it gets an immutable view of the speech frame and updates the noise level estimate 3. Gain applier - it gets the speech frame, the current speech and noise estimates, and the speech probability. It finds a gain to apply and applies it. 4. AdaptiveAgc - sets up and controls the sub-modules described above. Bug: webrtc:7494 Change-Id: Ib7ccd8924e94eead0bc5f935b5d8a12e06e24fd1 Reviewed-on: https://webrtc-review.googlesource.com/64440 Reviewed-by: Alessio Bazzica <alessiob@webrtc.org> Commit-Queue: Alex Loiko <aleloi@webrtc.org> Cr-Commit-Position: refs/heads/master@{#22628}
2018-03-27 13:38:36 +02:00
sources = [
AGC2 Bi-Quad filter: separate target and unit test. Adding a build target for the bi-qaud filter to make it available for the RNN VAD of AGC2. Also adding a unit test to test the computation both in-place and not in-place while comparing the produced output to that of scipy.signal. Bug: webrtc:9076 Change-Id: I16176a477ee4b81bb1e090c4906c3a9948ad2772 Reviewed-on: https://webrtc-review.googlesource.com/74220 Commit-Queue: Alessio Bazzica <alessiob@webrtc.org> Reviewed-by: Alex Loiko <aleloi@webrtc.org> Cr-Commit-Position: refs/heads/master@{#23141}
2018-05-07 13:26:47 +02:00
"gain_applier.cc",
"gain_applier.h",
Adaptive Digital gain control structure. This CL defines the control flow of the adaptive AGC. It also defines method and class stubs. Contents: 1. Divide the 'agc2' build target into 'fixed_digital' and 'adaptive_digital'. 1. Update the dependencies of everything that depended on 'agc2'. 2. Define the sub-modules of the adaptive digital AGC 2. They are: 1. Level Estimator - it gets the energy and a speech probability and updates a speech level estimate. 2. Noise Estimator - it gets an immutable view of the speech frame and updates the noise level estimate 3. Gain applier - it gets the speech frame, the current speech and noise estimates, and the speech probability. It finds a gain to apply and applies it. 4. AdaptiveAgc - sets up and controls the sub-modules described above. Bug: webrtc:7494 Change-Id: Ib7ccd8924e94eead0bc5f935b5d8a12e06e24fd1 Reviewed-on: https://webrtc-review.googlesource.com/64440 Reviewed-by: Alessio Bazzica <alessiob@webrtc.org> Commit-Queue: Alex Loiko <aleloi@webrtc.org> Cr-Commit-Position: refs/heads/master@{#22628}
2018-03-27 13:38:36 +02:00
]
deps = [
AGC2 Bi-Quad filter: separate target and unit test. Adding a build target for the bi-qaud filter to make it available for the RNN VAD of AGC2. Also adding a unit test to test the computation both in-place and not in-place while comparing the produced output to that of scipy.signal. Bug: webrtc:9076 Change-Id: I16176a477ee4b81bb1e090c4906c3a9948ad2772 Reviewed-on: https://webrtc-review.googlesource.com/74220 Commit-Queue: Alessio Bazzica <alessiob@webrtc.org> Reviewed-by: Alex Loiko <aleloi@webrtc.org> Cr-Commit-Position: refs/heads/master@{#23141}
2018-05-07 13:26:47 +02:00
":common",
"..:audio_frame_view",
"../../../rtc_base:safe_minmax",
Adaptive Digital gain control structure. This CL defines the control flow of the adaptive AGC. It also defines method and class stubs. Contents: 1. Divide the 'agc2' build target into 'fixed_digital' and 'adaptive_digital'. 1. Update the dependencies of everything that depended on 'agc2'. 2. Define the sub-modules of the adaptive digital AGC 2. They are: 1. Level Estimator - it gets the energy and a speech probability and updates a speech level estimate. 2. Noise Estimator - it gets an immutable view of the speech frame and updates the noise level estimate 3. Gain applier - it gets the speech frame, the current speech and noise estimates, and the speech probability. It finds a gain to apply and applies it. 4. AdaptiveAgc - sets up and controls the sub-modules described above. Bug: webrtc:7494 Change-Id: Ib7ccd8924e94eead0bc5f935b5d8a12e06e24fd1 Reviewed-on: https://webrtc-review.googlesource.com/64440 Reviewed-by: Alessio Bazzica <alessiob@webrtc.org> Commit-Queue: Alex Loiko <aleloi@webrtc.org> Cr-Commit-Position: refs/heads/master@{#22628}
2018-03-27 13:38:36 +02:00
]
}
Noise level estimation for AGC2. We put back the old noise estimator from LevelController. We add a few new unit tests. We also re-arrange the code so that it fits with how it is used in AGC2. The differences are: 1. The NoiseLevelEstimator is now fully self-contained. 2. The NoiseLevelEstimator is responsible for calling SignalClassifier and computing the signal energy. Previously the signal type and energy were used in several places. It made sense to compute the values independently of the noise calculation. 3. Re-initialization doesn't have to be done by the caller. 4. The interface is AudioFrameView instead of AudioBuffer. # Bots are green, nothing should break internal stuff NOTRY=True Bug: webrtc:7494 Change-Id: I442bdbbeb3796eb2518e96000aec9dc5a039ae6d Reviewed-on: https://webrtc-review.googlesource.com/66380 Commit-Queue: Alex Loiko <aleloi@webrtc.org> Reviewed-by: Sam Zackrisson <saza@webrtc.org> Cr-Commit-Position: refs/heads/master@{#22738}
2018-04-04 15:05:57 +02:00
rtc_source_set("noise_level_estimator") {
sources = [
"down_sampler.cc",
"down_sampler.h",
"noise_level_estimator.cc",
"noise_level_estimator.h",
"noise_spectrum_estimator.cc",
"noise_spectrum_estimator.h",
"signal_classifier.cc",
"signal_classifier.h",
]
deps = [
AGC2 Bi-Quad filter: separate target and unit test. Adding a build target for the bi-qaud filter to make it available for the RNN VAD of AGC2. Also adding a unit test to test the computation both in-place and not in-place while comparing the produced output to that of scipy.signal. Bug: webrtc:9076 Change-Id: I16176a477ee4b81bb1e090c4906c3a9948ad2772 Reviewed-on: https://webrtc-review.googlesource.com/74220 Commit-Queue: Alessio Bazzica <alessiob@webrtc.org> Reviewed-by: Alex Loiko <aleloi@webrtc.org> Cr-Commit-Position: refs/heads/master@{#23141}
2018-05-07 13:26:47 +02:00
":biquad_filter",
Noise level estimation for AGC2. We put back the old noise estimator from LevelController. We add a few new unit tests. We also re-arrange the code so that it fits with how it is used in AGC2. The differences are: 1. The NoiseLevelEstimator is now fully self-contained. 2. The NoiseLevelEstimator is responsible for calling SignalClassifier and computing the signal energy. Previously the signal type and energy were used in several places. It made sense to compute the values independently of the noise calculation. 3. Re-initialization doesn't have to be done by the caller. 4. The interface is AudioFrameView instead of AudioBuffer. # Bots are green, nothing should break internal stuff NOTRY=True Bug: webrtc:7494 Change-Id: I442bdbbeb3796eb2518e96000aec9dc5a039ae6d Reviewed-on: https://webrtc-review.googlesource.com/66380 Commit-Queue: Alex Loiko <aleloi@webrtc.org> Reviewed-by: Sam Zackrisson <saza@webrtc.org> Cr-Commit-Position: refs/heads/master@{#22738}
2018-04-04 15:05:57 +02:00
"..:apm_logging",
"..:audio_frame_view",
"../../../api:array_view",
"../../../common_audio",
"../../../rtc_base:checks",
"../../../rtc_base:macromagic",
Create separate build targets for utility/ in APM This clarifies the dependencies of utility/ a lot (spoiler: there are very few) and makes it easier to separate the build targets for aecm and aec2. Bug: webrtc:9488 Change-Id: If916f86e80c19d1b650d0908fbe8343ea7c47bd7 Reviewed-on: https://webrtc-review.googlesource.com/87141 Reviewed-by: Alex Loiko <aleloi@webrtc.org> Commit-Queue: Sam Zackrisson <saza@webrtc.org> Cr-Commit-Position: refs/heads/master@{#23860}
2018-07-05 16:41:55 +02:00
"../utility:ooura_fft",
Noise level estimation for AGC2. We put back the old noise estimator from LevelController. We add a few new unit tests. We also re-arrange the code so that it fits with how it is used in AGC2. The differences are: 1. The NoiseLevelEstimator is now fully self-contained. 2. The NoiseLevelEstimator is responsible for calling SignalClassifier and computing the signal energy. Previously the signal type and energy were used in several places. It made sense to compute the values independently of the noise calculation. 3. Re-initialization doesn't have to be done by the caller. 4. The interface is AudioFrameView instead of AudioBuffer. # Bots are green, nothing should break internal stuff NOTRY=True Bug: webrtc:7494 Change-Id: I442bdbbeb3796eb2518e96000aec9dc5a039ae6d Reviewed-on: https://webrtc-review.googlesource.com/66380 Commit-Queue: Alex Loiko <aleloi@webrtc.org> Reviewed-by: Sam Zackrisson <saza@webrtc.org> Cr-Commit-Position: refs/heads/master@{#22738}
2018-04-04 15:05:57 +02:00
]
configs += [ "..:apm_debug_dump" ]
}
Add RNN-VAD to AGC2. * Move 'VadWithLevel' to AGC2 where it belongs. * Remove the vectors from VadWithLevel. They were there to make it work with modules/audio_processing/vad, which we don't need any longer. * Remove the vector handling from AGC2. It was spread out across AdaptiveDigitalGainApplier, AdaptiveAGC and their unit tests. * Hack the RNN VAD into VadWithLevel. The main issue is the resampling. Bug: webrtc:9076 Change-Id: I13056c985d0ec41269735150caf4aaeb6ff9281e Reviewed-on: https://webrtc-review.googlesource.com/77364 Reviewed-by: Sam Zackrisson <saza@webrtc.org> Commit-Queue: Alex Loiko <aleloi@webrtc.org> Cr-Commit-Position: refs/heads/master@{#23688}
2018-06-20 14:14:18 +02:00
rtc_source_set("rnn_vad_with_level") {
sources = [
"vad_with_level.cc",
"vad_with_level.h",
]
deps = [
"..:audio_frame_view",
"../../../api:array_view",
"../../../common_audio",
"../../../rtc_base:checks",
RNN VAD: more specific build target names. Bug: webrtc:9076 Change-Id: Ie35ce0f864318a1ddc552285a5535fe411168202 Reviewed-on: https://webrtc-review.googlesource.com/91760 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Per Åhgren <peah@webrtc.org> Commit-Queue: Alessio Bazzica <alessiob@webrtc.org> Cr-Commit-Position: refs/heads/master@{#24162}
2018-08-01 10:24:15 +02:00
"rnn_vad",
Add RNN-VAD to AGC2. * Move 'VadWithLevel' to AGC2 where it belongs. * Remove the vectors from VadWithLevel. They were there to make it work with modules/audio_processing/vad, which we don't need any longer. * Remove the vector handling from AGC2. It was spread out across AdaptiveDigitalGainApplier, AdaptiveAGC and their unit tests. * Hack the RNN VAD into VadWithLevel. The main issue is the resampling. Bug: webrtc:9076 Change-Id: I13056c985d0ec41269735150caf4aaeb6ff9281e Reviewed-on: https://webrtc-review.googlesource.com/77364 Reviewed-by: Sam Zackrisson <saza@webrtc.org> Commit-Queue: Alex Loiko <aleloi@webrtc.org> Cr-Commit-Position: refs/heads/master@{#23688}
2018-06-20 14:14:18 +02:00
]
}
AGC2 Bi-Quad filter: separate target and unit test. Adding a build target for the bi-qaud filter to make it available for the RNN VAD of AGC2. Also adding a unit test to test the computation both in-place and not in-place while comparing the produced output to that of scipy.signal. Bug: webrtc:9076 Change-Id: I16176a477ee4b81bb1e090c4906c3a9948ad2772 Reviewed-on: https://webrtc-review.googlesource.com/74220 Commit-Queue: Alessio Bazzica <alessiob@webrtc.org> Reviewed-by: Alex Loiko <aleloi@webrtc.org> Cr-Commit-Position: refs/heads/master@{#23141}
2018-05-07 13:26:47 +02:00
rtc_source_set("adaptive_digital_unittests") {
testonly = true
configs += [ "..:apm_debug_dump" ]
Pre-amplification in the audio processing module. Added a new sub-module 'GainApplier'. The build target is 'modules/audio_processing/agc2:gain_applier'. A small refactoring makes the GainApplier used in adaptive-digital AGC2. The AGC2 now multiplies samples with a gain in 3 places. It's the GainApplier, the GainCurveApplier, and the FixedGainController. The GainApplier is used in AdaptiveDigitalGainApplier and will be used as a pre-amplifier. Bug: webrtc:9138 Change-Id: Ibc4c0ea109c6757f159d4adb6e3d8614179c9bc6 Reviewed-on: https://webrtc-review.googlesource.com/69321 Commit-Queue: Alex Loiko <aleloi@webrtc.org> Reviewed-by: Alessio Bazzica <alessiob@webrtc.org> Cr-Commit-Position: refs/heads/master@{#22849}
2018-04-13 11:15:34 +02:00
sources = [
AGC2 Bi-Quad filter: separate target and unit test. Adding a build target for the bi-qaud filter to make it available for the RNN VAD of AGC2. Also adding a unit test to test the computation both in-place and not in-place while comparing the produced output to that of scipy.signal. Bug: webrtc:9076 Change-Id: I16176a477ee4b81bb1e090c4906c3a9948ad2772 Reviewed-on: https://webrtc-review.googlesource.com/74220 Commit-Queue: Alessio Bazzica <alessiob@webrtc.org> Reviewed-by: Alex Loiko <aleloi@webrtc.org> Cr-Commit-Position: refs/heads/master@{#23141}
2018-05-07 13:26:47 +02:00
"adaptive_digital_gain_applier_unittest.cc",
"adaptive_mode_level_estimator_unittest.cc",
"gain_applier_unittest.cc",
"saturation_protector_unittest.cc",
Pre-amplification in the audio processing module. Added a new sub-module 'GainApplier'. The build target is 'modules/audio_processing/agc2:gain_applier'. A small refactoring makes the GainApplier used in adaptive-digital AGC2. The AGC2 now multiplies samples with a gain in 3 places. It's the GainApplier, the GainCurveApplier, and the FixedGainController. The GainApplier is used in AdaptiveDigitalGainApplier and will be used as a pre-amplifier. Bug: webrtc:9138 Change-Id: Ibc4c0ea109c6757f159d4adb6e3d8614179c9bc6 Reviewed-on: https://webrtc-review.googlesource.com/69321 Commit-Queue: Alex Loiko <aleloi@webrtc.org> Reviewed-by: Alessio Bazzica <alessiob@webrtc.org> Cr-Commit-Position: refs/heads/master@{#22849}
2018-04-13 11:15:34 +02:00
]
deps = [
AGC2 Bi-Quad filter: separate target and unit test. Adding a build target for the bi-qaud filter to make it available for the RNN VAD of AGC2. Also adding a unit test to test the computation both in-place and not in-place while comparing the produced output to that of scipy.signal. Bug: webrtc:9076 Change-Id: I16176a477ee4b81bb1e090c4906c3a9948ad2772 Reviewed-on: https://webrtc-review.googlesource.com/74220 Commit-Queue: Alessio Bazzica <alessiob@webrtc.org> Reviewed-by: Alex Loiko <aleloi@webrtc.org> Cr-Commit-Position: refs/heads/master@{#23141}
2018-05-07 13:26:47 +02:00
":adaptive_digital",
Pre-amplification in the audio processing module. Added a new sub-module 'GainApplier'. The build target is 'modules/audio_processing/agc2:gain_applier'. A small refactoring makes the GainApplier used in adaptive-digital AGC2. The AGC2 now multiplies samples with a gain in 3 places. It's the GainApplier, the GainCurveApplier, and the FixedGainController. The GainApplier is used in AdaptiveDigitalGainApplier and will be used as a pre-amplifier. Bug: webrtc:9138 Change-Id: Ibc4c0ea109c6757f159d4adb6e3d8614179c9bc6 Reviewed-on: https://webrtc-review.googlesource.com/69321 Commit-Queue: Alex Loiko <aleloi@webrtc.org> Reviewed-by: Alessio Bazzica <alessiob@webrtc.org> Cr-Commit-Position: refs/heads/master@{#22849}
2018-04-13 11:15:34 +02:00
":common",
AGC2 Bi-Quad filter: separate target and unit test. Adding a build target for the bi-qaud filter to make it available for the RNN VAD of AGC2. Also adding a unit test to test the computation both in-place and not in-place while comparing the produced output to that of scipy.signal. Bug: webrtc:9076 Change-Id: I16176a477ee4b81bb1e090c4906c3a9948ad2772 Reviewed-on: https://webrtc-review.googlesource.com/74220 Commit-Queue: Alessio Bazzica <alessiob@webrtc.org> Reviewed-by: Alex Loiko <aleloi@webrtc.org> Cr-Commit-Position: refs/heads/master@{#23141}
2018-05-07 13:26:47 +02:00
":gain_applier",
":test_utils",
"..:apm_logging",
Pre-amplification in the audio processing module. Added a new sub-module 'GainApplier'. The build target is 'modules/audio_processing/agc2:gain_applier'. A small refactoring makes the GainApplier used in adaptive-digital AGC2. The AGC2 now multiplies samples with a gain in 3 places. It's the GainApplier, the GainCurveApplier, and the FixedGainController. The GainApplier is used in AdaptiveDigitalGainApplier and will be used as a pre-amplifier. Bug: webrtc:9138 Change-Id: Ibc4c0ea109c6757f159d4adb6e3d8614179c9bc6 Reviewed-on: https://webrtc-review.googlesource.com/69321 Commit-Queue: Alex Loiko <aleloi@webrtc.org> Reviewed-by: Alessio Bazzica <alessiob@webrtc.org> Cr-Commit-Position: refs/heads/master@{#22849}
2018-04-13 11:15:34 +02:00
"..:audio_frame_view",
AGC2 Bi-Quad filter: separate target and unit test. Adding a build target for the bi-qaud filter to make it available for the RNN VAD of AGC2. Also adding a unit test to test the computation both in-place and not in-place while comparing the produced output to that of scipy.signal. Bug: webrtc:9076 Change-Id: I16176a477ee4b81bb1e090c4906c3a9948ad2772 Reviewed-on: https://webrtc-review.googlesource.com/74220 Commit-Queue: Alessio Bazzica <alessiob@webrtc.org> Reviewed-by: Alex Loiko <aleloi@webrtc.org> Cr-Commit-Position: refs/heads/master@{#23141}
2018-05-07 13:26:47 +02:00
"../../../api:array_view",
"../../../common_audio",
"../../../rtc_base:checks",
"../../../rtc_base:rtc_base_approved",
"../../../rtc_base:rtc_base_tests_utils",
Pre-amplification in the audio processing module. Added a new sub-module 'GainApplier'. The build target is 'modules/audio_processing/agc2:gain_applier'. A small refactoring makes the GainApplier used in adaptive-digital AGC2. The AGC2 now multiplies samples with a gain in 3 places. It's the GainApplier, the GainCurveApplier, and the FixedGainController. The GainApplier is used in AdaptiveDigitalGainApplier and will be used as a pre-amplifier. Bug: webrtc:9138 Change-Id: Ibc4c0ea109c6757f159d4adb6e3d8614179c9bc6 Reviewed-on: https://webrtc-review.googlesource.com/69321 Commit-Queue: Alex Loiko <aleloi@webrtc.org> Reviewed-by: Alessio Bazzica <alessiob@webrtc.org> Cr-Commit-Position: refs/heads/master@{#22849}
2018-04-13 11:15:34 +02:00
]
}
AGC2 Bi-Quad filter: separate target and unit test. Adding a build target for the bi-qaud filter to make it available for the RNN VAD of AGC2. Also adding a unit test to test the computation both in-place and not in-place while comparing the produced output to that of scipy.signal. Bug: webrtc:9076 Change-Id: I16176a477ee4b81bb1e090c4906c3a9948ad2772 Reviewed-on: https://webrtc-review.googlesource.com/74220 Commit-Queue: Alessio Bazzica <alessiob@webrtc.org> Reviewed-by: Alex Loiko <aleloi@webrtc.org> Cr-Commit-Position: refs/heads/master@{#23141}
2018-05-07 13:26:47 +02:00
rtc_source_set("biquad_filter_unittests") {
Adaptive Digital gain control structure. This CL defines the control flow of the adaptive AGC. It also defines method and class stubs. Contents: 1. Divide the 'agc2' build target into 'fixed_digital' and 'adaptive_digital'. 1. Update the dependencies of everything that depended on 'agc2'. 2. Define the sub-modules of the adaptive digital AGC 2. They are: 1. Level Estimator - it gets the energy and a speech probability and updates a speech level estimate. 2. Noise Estimator - it gets an immutable view of the speech frame and updates the noise level estimate 3. Gain applier - it gets the speech frame, the current speech and noise estimates, and the speech probability. It finds a gain to apply and applies it. 4. AdaptiveAgc - sets up and controls the sub-modules described above. Bug: webrtc:7494 Change-Id: Ib7ccd8924e94eead0bc5f935b5d8a12e06e24fd1 Reviewed-on: https://webrtc-review.googlesource.com/64440 Reviewed-by: Alessio Bazzica <alessiob@webrtc.org> Commit-Queue: Alex Loiko <aleloi@webrtc.org> Cr-Commit-Position: refs/heads/master@{#22628}
2018-03-27 13:38:36 +02:00
testonly = true
sources = [
AGC2 Bi-Quad filter: separate target and unit test. Adding a build target for the bi-qaud filter to make it available for the RNN VAD of AGC2. Also adding a unit test to test the computation both in-place and not in-place while comparing the produced output to that of scipy.signal. Bug: webrtc:9076 Change-Id: I16176a477ee4b81bb1e090c4906c3a9948ad2772 Reviewed-on: https://webrtc-review.googlesource.com/74220 Commit-Queue: Alessio Bazzica <alessiob@webrtc.org> Reviewed-by: Alex Loiko <aleloi@webrtc.org> Cr-Commit-Position: refs/heads/master@{#23141}
2018-05-07 13:26:47 +02:00
"biquad_filter_unittest.cc",
Adaptive Digital gain control structure. This CL defines the control flow of the adaptive AGC. It also defines method and class stubs. Contents: 1. Divide the 'agc2' build target into 'fixed_digital' and 'adaptive_digital'. 1. Update the dependencies of everything that depended on 'agc2'. 2. Define the sub-modules of the adaptive digital AGC 2. They are: 1. Level Estimator - it gets the energy and a speech probability and updates a speech level estimate. 2. Noise Estimator - it gets an immutable view of the speech frame and updates the noise level estimate 3. Gain applier - it gets the speech frame, the current speech and noise estimates, and the speech probability. It finds a gain to apply and applies it. 4. AdaptiveAgc - sets up and controls the sub-modules described above. Bug: webrtc:7494 Change-Id: Ib7ccd8924e94eead0bc5f935b5d8a12e06e24fd1 Reviewed-on: https://webrtc-review.googlesource.com/64440 Reviewed-by: Alessio Bazzica <alessiob@webrtc.org> Commit-Queue: Alex Loiko <aleloi@webrtc.org> Cr-Commit-Position: refs/heads/master@{#22628}
2018-03-27 13:38:36 +02:00
]
deps = [
AGC2 Bi-Quad filter: separate target and unit test. Adding a build target for the bi-qaud filter to make it available for the RNN VAD of AGC2. Also adding a unit test to test the computation both in-place and not in-place while comparing the produced output to that of scipy.signal. Bug: webrtc:9076 Change-Id: I16176a477ee4b81bb1e090c4906c3a9948ad2772 Reviewed-on: https://webrtc-review.googlesource.com/74220 Commit-Queue: Alessio Bazzica <alessiob@webrtc.org> Reviewed-by: Alex Loiko <aleloi@webrtc.org> Cr-Commit-Position: refs/heads/master@{#23141}
2018-05-07 13:26:47 +02:00
":biquad_filter",
"../../../rtc_base:rtc_base_tests_utils",
Adaptive Digital gain control structure. This CL defines the control flow of the adaptive AGC. It also defines method and class stubs. Contents: 1. Divide the 'agc2' build target into 'fixed_digital' and 'adaptive_digital'. 1. Update the dependencies of everything that depended on 'agc2'. 2. Define the sub-modules of the adaptive digital AGC 2. They are: 1. Level Estimator - it gets the energy and a speech probability and updates a speech level estimate. 2. Noise Estimator - it gets an immutable view of the speech frame and updates the noise level estimate 3. Gain applier - it gets the speech frame, the current speech and noise estimates, and the speech probability. It finds a gain to apply and applies it. 4. AdaptiveAgc - sets up and controls the sub-modules described above. Bug: webrtc:7494 Change-Id: Ib7ccd8924e94eead0bc5f935b5d8a12e06e24fd1 Reviewed-on: https://webrtc-review.googlesource.com/64440 Reviewed-by: Alessio Bazzica <alessiob@webrtc.org> Commit-Queue: Alex Loiko <aleloi@webrtc.org> Cr-Commit-Position: refs/heads/master@{#22628}
2018-03-27 13:38:36 +02:00
]
}
Add FixedGainController and move GainController2 in APM. 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 11:54:07 +01:00
rtc_source_set("fixed_digital_unittests") {
testonly = true
configs += [ "..:apm_debug_dump" ]
sources = [
Fixed Digital mode of AGC2 implementation finished. This CL adds the GainCurveApplier (GCA). It owns a FixedDigitalLevelEstimator (LE) and an InterpolatedGainCurve (IGC). The GCA uses the LE to compute the input signal level, looks up a gain from IGC and applies it on the signal. The other IGC and LE submodules were added in previous CLs [1] and [2]. This CL also turns on AGC2 in the APM fuzzer. [1] https://webrtc-review.googlesource.com/c/src/+/51920 [2] https://webrtc-review.googlesource.com/c/src/+/52381 Bug: webrtc:7949 Change-Id: Idb10cc3ca9d6d2e4ac5824cc3391ed8aa680f6cd Reviewed-on: https://webrtc-review.googlesource.com/54361 Commit-Queue: Alex Loiko <aleloi@webrtc.org> Reviewed-by: Sam Zackrisson <saza@webrtc.org> Cr-Commit-Position: refs/heads/master@{#22103}
2018-02-20 15:58:36 +01:00
"agc2_testing_common_unittest.cc",
"compute_interpolated_gain_curve.cc",
"compute_interpolated_gain_curve.h",
AGC2-fixed-digital: Level Estimator This CL adds the Level Estimator of the new gain controller. The Level Estimator divides a 10ms input frame in kSubFramesInFrame=20 sub frames. We take the maximal sample values in every sub frame. We then apply attack/decay smoothing. This is the final level estimate. The results will be used with InterpolatedGainCurve (see this CL https://webrtc-review.googlesource.com/c/src/+/51920). For every level estimate value, we look up a gain with InterpolatedGainCurve::LookUpGainToApply. This gain is then applied to the signal. Bug: webrtc:7949 Change-Id: I2b4b3894a3e945d3dd916ce516c79abacb2b18b1 Reviewed-on: https://webrtc-review.googlesource.com/52381 Reviewed-by: Sam Zackrisson <saza@webrtc.org> Commit-Queue: Alex Loiko <aleloi@webrtc.org> Cr-Commit-Position: refs/heads/master@{#22054}
2018-02-16 12:39:00 +01:00
"fixed_digital_level_estimator_unittest.cc",
Add FixedGainController and move GainController2 in APM. 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 11:54:07 +01:00
"fixed_gain_controller_unittest.cc",
Fixed Digital mode of AGC2 implementation finished. This CL adds the GainCurveApplier (GCA). It owns a FixedDigitalLevelEstimator (LE) and an InterpolatedGainCurve (IGC). The GCA uses the LE to compute the input signal level, looks up a gain from IGC and applies it on the signal. The other IGC and LE submodules were added in previous CLs [1] and [2]. This CL also turns on AGC2 in the APM fuzzer. [1] https://webrtc-review.googlesource.com/c/src/+/51920 [2] https://webrtc-review.googlesource.com/c/src/+/52381 Bug: webrtc:7949 Change-Id: Idb10cc3ca9d6d2e4ac5824cc3391ed8aa680f6cd Reviewed-on: https://webrtc-review.googlesource.com/54361 Commit-Queue: Alex Loiko <aleloi@webrtc.org> Reviewed-by: Sam Zackrisson <saza@webrtc.org> Cr-Commit-Position: refs/heads/master@{#22103}
2018-02-20 15:58:36 +01:00
"gain_curve_applier_unittest.cc",
"interpolated_gain_curve_unittest.cc",
"limiter.cc",
"limiter.h",
"limiter_unittest.cc",
Add FixedGainController and move GainController2 in APM. 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 11:54:07 +01:00
]
deps = [
Adaptive Digital gain control structure. This CL defines the control flow of the adaptive AGC. It also defines method and class stubs. Contents: 1. Divide the 'agc2' build target into 'fixed_digital' and 'adaptive_digital'. 1. Update the dependencies of everything that depended on 'agc2'. 2. Define the sub-modules of the adaptive digital AGC 2. They are: 1. Level Estimator - it gets the energy and a speech probability and updates a speech level estimate. 2. Noise Estimator - it gets an immutable view of the speech frame and updates the noise level estimate 3. Gain applier - it gets the speech frame, the current speech and noise estimates, and the speech probability. It finds a gain to apply and applies it. 4. AdaptiveAgc - sets up and controls the sub-modules described above. Bug: webrtc:7494 Change-Id: Ib7ccd8924e94eead0bc5f935b5d8a12e06e24fd1 Reviewed-on: https://webrtc-review.googlesource.com/64440 Reviewed-by: Alessio Bazzica <alessiob@webrtc.org> Commit-Queue: Alex Loiko <aleloi@webrtc.org> Cr-Commit-Position: refs/heads/master@{#22628}
2018-03-27 13:38:36 +02:00
":common",
":fixed_digital",
":test_utils",
Add FixedGainController and move GainController2 in APM. 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 11:54:07 +01:00
"..:apm_logging",
"..:audio_frame_view",
"../../../api:array_view",
AGC2-fixed-digital: Level Estimator This CL adds the Level Estimator of the new gain controller. The Level Estimator divides a 10ms input frame in kSubFramesInFrame=20 sub frames. We take the maximal sample values in every sub frame. We then apply attack/decay smoothing. This is the final level estimate. The results will be used with InterpolatedGainCurve (see this CL https://webrtc-review.googlesource.com/c/src/+/51920). For every level estimate value, we look up a gain with InterpolatedGainCurve::LookUpGainToApply. This gain is then applied to the signal. Bug: webrtc:7949 Change-Id: I2b4b3894a3e945d3dd916ce516c79abacb2b18b1 Reviewed-on: https://webrtc-review.googlesource.com/52381 Reviewed-by: Sam Zackrisson <saza@webrtc.org> Commit-Queue: Alex Loiko <aleloi@webrtc.org> Cr-Commit-Position: refs/heads/master@{#22054}
2018-02-16 12:39:00 +01:00
"../../../common_audio",
Fixed Digital mode of AGC2 implementation finished. This CL adds the GainCurveApplier (GCA). It owns a FixedDigitalLevelEstimator (LE) and an InterpolatedGainCurve (IGC). The GCA uses the LE to compute the input signal level, looks up a gain from IGC and applies it on the signal. The other IGC and LE submodules were added in previous CLs [1] and [2]. This CL also turns on AGC2 in the APM fuzzer. [1] https://webrtc-review.googlesource.com/c/src/+/51920 [2] https://webrtc-review.googlesource.com/c/src/+/52381 Bug: webrtc:7949 Change-Id: Idb10cc3ca9d6d2e4ac5824cc3391ed8aa680f6cd Reviewed-on: https://webrtc-review.googlesource.com/54361 Commit-Queue: Alex Loiko <aleloi@webrtc.org> Reviewed-by: Sam Zackrisson <saza@webrtc.org> Cr-Commit-Position: refs/heads/master@{#22103}
2018-02-20 15:58:36 +01:00
"../../../rtc_base:checks",
"../../../rtc_base:rtc_base_approved",
Add FixedGainController and move GainController2 in APM. 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 11:54:07 +01:00
"../../../rtc_base:rtc_base_tests_utils",
Fine-grained limiter metrics. The FixedGainController is used in two places. One is the AudioMixer. There it's used to limit the audio level after adding streams. The other is GainController2, where it's placed after steps that could boost the audio level outside the allowed range. We log metrics from the FGC. To avoid confusion, this CL makes the two use cases log to different histograms. Chromium histogram CL is https://chromium-review.googlesource.com/c/chromium/src/+/1170833 Bug: webrtc:7494 Change-Id: I1abe60fd8e96556f144d2ee576254b15beca1174 Reviewed-on: https://webrtc-review.googlesource.com/93464 Commit-Queue: Alex Loiko <aleloi@webrtc.org> Reviewed-by: Ivo Creusen <ivoc@webrtc.org> Reviewed-by: Åsa Persson <asapersson@webrtc.org> Cr-Commit-Position: refs/heads/master@{#24284}
2018-08-13 17:40:43 +02:00
"../../../system_wrappers:metrics_default",
Use absl::make_unique and absl::WrapUnique directly Instead of going through our wrappers in ptr_util.h. This CL was generated by the following script: git grep -l ptr_util | xargs perl -pi -e 's,#include "rtc_base/ptr_util.h",#include "absl/memory/memory.h",' git grep -l MakeUnique | xargs perl -pi -e 's,\b(rtc::)?MakeUnique\b,absl::make_unique,g' git grep -l WrapUnique | xargs perl -pi -e 's,\b(rtc::)?WrapUnique\b,absl::WrapUnique,g' git checkout -- rtc_base/ptr_util{.h,_unittest.cc} git cl format Followed by manually adding dependencies on //third_party/abseil-cpp/absl/memory until `gn check` stopped complaining. Bug: webrtc:9473 Change-Id: I89ccd363f070479b8c431eb2c3d404a46eaacc1c Reviewed-on: https://webrtc-review.googlesource.com/86600 Commit-Queue: Karl Wiberg <kwiberg@webrtc.org> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/master@{#23850}
2018-07-05 11:40:33 +02:00
"//third_party/abseil-cpp/absl/memory",
Add FixedGainController and move GainController2 in APM. 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 11:54:07 +01:00
]
}
Level estimation and saturation protection stub. The level estimator (AdaptiveModeLevelEstimator) produces a biased estimate of the speech level. In our model, we use another module (the SaturationProtector) to compute the bias. This CL contains the estimator and a stub of the saturation protector. Bug: webrtc:7494 Change-Id: I0df736d0346063f544fa680b4cc84177ea548545 Reviewed-on: https://webrtc-review.googlesource.com/64820 Commit-Queue: Alex Loiko <aleloi@webrtc.org> Reviewed-by: Ivo Creusen <ivoc@webrtc.org> Cr-Commit-Position: refs/heads/master@{#22641}
2018-03-28 09:45:29 +02:00
AGC2 Bi-Quad filter: separate target and unit test. Adding a build target for the bi-qaud filter to make it available for the RNN VAD of AGC2. Also adding a unit test to test the computation both in-place and not in-place while comparing the produced output to that of scipy.signal. Bug: webrtc:9076 Change-Id: I16176a477ee4b81bb1e090c4906c3a9948ad2772 Reviewed-on: https://webrtc-review.googlesource.com/74220 Commit-Queue: Alessio Bazzica <alessiob@webrtc.org> Reviewed-by: Alex Loiko <aleloi@webrtc.org> Cr-Commit-Position: refs/heads/master@{#23141}
2018-05-07 13:26:47 +02:00
rtc_source_set("noise_estimator_unittests") {
Level estimation and saturation protection stub. The level estimator (AdaptiveModeLevelEstimator) produces a biased estimate of the speech level. In our model, we use another module (the SaturationProtector) to compute the bias. This CL contains the estimator and a stub of the saturation protector. Bug: webrtc:7494 Change-Id: I0df736d0346063f544fa680b4cc84177ea548545 Reviewed-on: https://webrtc-review.googlesource.com/64820 Commit-Queue: Alex Loiko <aleloi@webrtc.org> Reviewed-by: Ivo Creusen <ivoc@webrtc.org> Cr-Commit-Position: refs/heads/master@{#22641}
2018-03-28 09:45:29 +02:00
testonly = true
configs += [ "..:apm_debug_dump" ]
sources = [
AGC2 Bi-Quad filter: separate target and unit test. Adding a build target for the bi-qaud filter to make it available for the RNN VAD of AGC2. Also adding a unit test to test the computation both in-place and not in-place while comparing the produced output to that of scipy.signal. Bug: webrtc:9076 Change-Id: I16176a477ee4b81bb1e090c4906c3a9948ad2772 Reviewed-on: https://webrtc-review.googlesource.com/74220 Commit-Queue: Alessio Bazzica <alessiob@webrtc.org> Reviewed-by: Alex Loiko <aleloi@webrtc.org> Cr-Commit-Position: refs/heads/master@{#23141}
2018-05-07 13:26:47 +02:00
"noise_level_estimator_unittest.cc",
"signal_classifier_unittest.cc",
Level estimation and saturation protection stub. The level estimator (AdaptiveModeLevelEstimator) produces a biased estimate of the speech level. In our model, we use another module (the SaturationProtector) to compute the bias. This CL contains the estimator and a stub of the saturation protector. Bug: webrtc:7494 Change-Id: I0df736d0346063f544fa680b4cc84177ea548545 Reviewed-on: https://webrtc-review.googlesource.com/64820 Commit-Queue: Alex Loiko <aleloi@webrtc.org> Reviewed-by: Ivo Creusen <ivoc@webrtc.org> Cr-Commit-Position: refs/heads/master@{#22641}
2018-03-28 09:45:29 +02:00
]
deps = [
AGC2 Bi-Quad filter: separate target and unit test. Adding a build target for the bi-qaud filter to make it available for the RNN VAD of AGC2. Also adding a unit test to test the computation both in-place and not in-place while comparing the produced output to that of scipy.signal. Bug: webrtc:9076 Change-Id: I16176a477ee4b81bb1e090c4906c3a9948ad2772 Reviewed-on: https://webrtc-review.googlesource.com/74220 Commit-Queue: Alessio Bazzica <alessiob@webrtc.org> Reviewed-by: Alex Loiko <aleloi@webrtc.org> Cr-Commit-Position: refs/heads/master@{#23141}
2018-05-07 13:26:47 +02:00
":noise_level_estimator",
Level estimation and saturation protection stub. The level estimator (AdaptiveModeLevelEstimator) produces a biased estimate of the speech level. In our model, we use another module (the SaturationProtector) to compute the bias. This CL contains the estimator and a stub of the saturation protector. Bug: webrtc:7494 Change-Id: I0df736d0346063f544fa680b4cc84177ea548545 Reviewed-on: https://webrtc-review.googlesource.com/64820 Commit-Queue: Alex Loiko <aleloi@webrtc.org> Reviewed-by: Ivo Creusen <ivoc@webrtc.org> Cr-Commit-Position: refs/heads/master@{#22641}
2018-03-28 09:45:29 +02:00
":test_utils",
"..:apm_logging",
"..:audio_frame_view",
"../../../api:array_view",
"../../../rtc_base:checks",
"../../../rtc_base:rtc_base_approved",
"../../../rtc_base:rtc_base_tests_utils",
]
}
Noise level estimation for AGC2. We put back the old noise estimator from LevelController. We add a few new unit tests. We also re-arrange the code so that it fits with how it is used in AGC2. The differences are: 1. The NoiseLevelEstimator is now fully self-contained. 2. The NoiseLevelEstimator is responsible for calling SignalClassifier and computing the signal energy. Previously the signal type and energy were used in several places. It made sense to compute the values independently of the noise calculation. 3. Re-initialization doesn't have to be done by the caller. 4. The interface is AudioFrameView instead of AudioBuffer. # Bots are green, nothing should break internal stuff NOTRY=True Bug: webrtc:7494 Change-Id: I442bdbbeb3796eb2518e96000aec9dc5a039ae6d Reviewed-on: https://webrtc-review.googlesource.com/66380 Commit-Queue: Alex Loiko <aleloi@webrtc.org> Reviewed-by: Sam Zackrisson <saza@webrtc.org> Cr-Commit-Position: refs/heads/master@{#22738}
2018-04-04 15:05:57 +02:00
AGC2 Bi-Quad filter: separate target and unit test. Adding a build target for the bi-qaud filter to make it available for the RNN VAD of AGC2. Also adding a unit test to test the computation both in-place and not in-place while comparing the produced output to that of scipy.signal. Bug: webrtc:9076 Change-Id: I16176a477ee4b81bb1e090c4906c3a9948ad2772 Reviewed-on: https://webrtc-review.googlesource.com/74220 Commit-Queue: Alessio Bazzica <alessiob@webrtc.org> Reviewed-by: Alex Loiko <aleloi@webrtc.org> Cr-Commit-Position: refs/heads/master@{#23141}
2018-05-07 13:26:47 +02:00
rtc_source_set("test_utils") {
Noise level estimation for AGC2. We put back the old noise estimator from LevelController. We add a few new unit tests. We also re-arrange the code so that it fits with how it is used in AGC2. The differences are: 1. The NoiseLevelEstimator is now fully self-contained. 2. The NoiseLevelEstimator is responsible for calling SignalClassifier and computing the signal energy. Previously the signal type and energy were used in several places. It made sense to compute the values independently of the noise calculation. 3. Re-initialization doesn't have to be done by the caller. 4. The interface is AudioFrameView instead of AudioBuffer. # Bots are green, nothing should break internal stuff NOTRY=True Bug: webrtc:7494 Change-Id: I442bdbbeb3796eb2518e96000aec9dc5a039ae6d Reviewed-on: https://webrtc-review.googlesource.com/66380 Commit-Queue: Alex Loiko <aleloi@webrtc.org> Reviewed-by: Sam Zackrisson <saza@webrtc.org> Cr-Commit-Position: refs/heads/master@{#22738}
2018-04-04 15:05:57 +02:00
testonly = true
AGC2 Bi-Quad filter: separate target and unit test. Adding a build target for the bi-qaud filter to make it available for the RNN VAD of AGC2. Also adding a unit test to test the computation both in-place and not in-place while comparing the produced output to that of scipy.signal. Bug: webrtc:9076 Change-Id: I16176a477ee4b81bb1e090c4906c3a9948ad2772 Reviewed-on: https://webrtc-review.googlesource.com/74220 Commit-Queue: Alessio Bazzica <alessiob@webrtc.org> Reviewed-by: Alex Loiko <aleloi@webrtc.org> Cr-Commit-Position: refs/heads/master@{#23141}
2018-05-07 13:26:47 +02:00
visibility = [ ":*" ]
Noise level estimation for AGC2. We put back the old noise estimator from LevelController. We add a few new unit tests. We also re-arrange the code so that it fits with how it is used in AGC2. The differences are: 1. The NoiseLevelEstimator is now fully self-contained. 2. The NoiseLevelEstimator is responsible for calling SignalClassifier and computing the signal energy. Previously the signal type and energy were used in several places. It made sense to compute the values independently of the noise calculation. 3. Re-initialization doesn't have to be done by the caller. 4. The interface is AudioFrameView instead of AudioBuffer. # Bots are green, nothing should break internal stuff NOTRY=True Bug: webrtc:7494 Change-Id: I442bdbbeb3796eb2518e96000aec9dc5a039ae6d Reviewed-on: https://webrtc-review.googlesource.com/66380 Commit-Queue: Alex Loiko <aleloi@webrtc.org> Reviewed-by: Sam Zackrisson <saza@webrtc.org> Cr-Commit-Position: refs/heads/master@{#22738}
2018-04-04 15:05:57 +02:00
sources = [
AGC2 Bi-Quad filter: separate target and unit test. Adding a build target for the bi-qaud filter to make it available for the RNN VAD of AGC2. Also adding a unit test to test the computation both in-place and not in-place while comparing the produced output to that of scipy.signal. Bug: webrtc:9076 Change-Id: I16176a477ee4b81bb1e090c4906c3a9948ad2772 Reviewed-on: https://webrtc-review.googlesource.com/74220 Commit-Queue: Alessio Bazzica <alessiob@webrtc.org> Reviewed-by: Alex Loiko <aleloi@webrtc.org> Cr-Commit-Position: refs/heads/master@{#23141}
2018-05-07 13:26:47 +02:00
"agc2_testing_common.cc",
"agc2_testing_common.h",
"vector_float_frame.cc",
"vector_float_frame.h",
Noise level estimation for AGC2. We put back the old noise estimator from LevelController. We add a few new unit tests. We also re-arrange the code so that it fits with how it is used in AGC2. The differences are: 1. The NoiseLevelEstimator is now fully self-contained. 2. The NoiseLevelEstimator is responsible for calling SignalClassifier and computing the signal energy. Previously the signal type and energy were used in several places. It made sense to compute the values independently of the noise calculation. 3. Re-initialization doesn't have to be done by the caller. 4. The interface is AudioFrameView instead of AudioBuffer. # Bots are green, nothing should break internal stuff NOTRY=True Bug: webrtc:7494 Change-Id: I442bdbbeb3796eb2518e96000aec9dc5a039ae6d Reviewed-on: https://webrtc-review.googlesource.com/66380 Commit-Queue: Alex Loiko <aleloi@webrtc.org> Reviewed-by: Sam Zackrisson <saza@webrtc.org> Cr-Commit-Position: refs/heads/master@{#22738}
2018-04-04 15:05:57 +02:00
]
deps = [
"..:audio_frame_view",
"../../../rtc_base:checks",
"../../../rtc_base:rtc_base_approved",
]
}
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