admin/webrtc_m130
1
0
Fork 0
Code Issues Pull Requests Actions Packages Projects Releases Wiki Activity
webrtc_m130/modules/audio_processing/agc2/BUILD.gn

Ignoring revisions in .git-blame-ignore-revs. Click here to bypass and see the normal blame view.

312 lines
6.8 KiB
Plaintext
Raw Normal View History

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",
]
}
Use source_sets in component builds and static_library in release builds. Static libraries don't guarantee that an exported symbol gets linked into a shared library (and in order to support Chromium's component build mode, WebRTC needs to be linked as a shared library). Source sets always pass all the object files to the linker. On the flip side, source_sets link more object files in release builds and to avoid this, this CL introduces a the GN template "rtc_library" that expands to static_library during release builds and to source_set during component builds. See: https://gn.googlesource.com/gn/+/master/docs/reference.md#func_source_set Bug: webrtc:9419 Change-Id: I4667e820c2b3fcec417becbd2034acc13e4f04fe Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/157168 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Reviewed-by: Nico Weber <thakis@chromium.org> Cr-Commit-Position: refs/heads/master@{#29525}
2019-10-17 21:32:04 +02:00
rtc_library("adaptive_digital") {
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 = [
"adaptive_digital_gain_applier.cc",
"adaptive_digital_gain_applier.h",
AGC2: `AdaptiveAgc` ctor with sample rate and # of channels 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}
2021-10-26 14:08:23 +02:00
"adaptive_digital_gain_controller.cc",
"adaptive_digital_gain_controller.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
"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",
AGC2: retuning and large refactoring - Bug fix: the desired initial gain quickly dropped to 0 dB hence starting a call with a too low level - New tuning to make AGC2 more robust to VAD mistakes - Smarter max gain increase speed: to deal with an increased threshold of adjacent speech frames, the gain applier temporarily allows a faster gain increase to deal with a longer time spent waiting for enough speech frames in a row to be observed - Saturation protector isolated from `AdaptiveModeLevelEstimator` to simplify the unit tests for the latter (non bit-exact change) - AGC2 adaptive digital config: unnecessary params deprecated - Code readability improvements - Data dumps clean-up and better naming Bug: webrtc:7494 Change-Id: I4e36059bdf2566cc2a7e1a7e95b7430ba9ae9844 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/215140 Commit-Queue: Alessio Bazzica <alessiob@webrtc.org> Reviewed-by: Jesus de Vicente Pena <devicentepena@webrtc.org> Cr-Commit-Position: refs/heads/master@{#33736}
2021-04-14 19:09:17 +02:00
"saturation_protector_buffer.cc",
"saturation_protector_buffer.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
]
AGC2: `gain_controller2` target isolated Needed to restrict visibility. Bug: webrtc:7494 Change-Id: I58a609666ca04d785c6dd2ed19233b395a94b06c Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/234584 Commit-Queue: Alessio Bazzica <alessiob@webrtc.org> Reviewed-by: Hanna Silen <silen@webrtc.org> Cr-Commit-Position: refs/heads/main@{#35201}
2021-10-14 10:51:40 +02:00
visibility = [
"..:gain_controller2",
"./*",
]
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",
AGC2 RNN VAD: safe SIMD optimizations scheme + AVX2 kill switch In preparation for adding AVX2 code, a safe scheme to support different SIMD optimizations is added. Safety features: - AVX2 kill switch to stop using it even if supported by the architecture - struct indicating the available CPU features propagated from AGC2 to each component; in this way - better control over the unit tests - no need to propagate individual kill switches but just set to false features that are turned off Note that (i) this CL does not change the performance of the RNN VAD and (ii) no AVX2 optimization is added yet. Bug: webrtc:10480 Change-Id: I0e61f3311ecd140f38369cf68b6e5954f3dc1f5a Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/193140 Reviewed-by: Per Åhgren <peah@webrtc.org> Commit-Queue: Alessio Bazzica <alessiob@webrtc.org> Cr-Commit-Position: refs/heads/master@{#32739}
2020-11-27 16:02:38 +01:00
":cpu_features",
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",
AGC2: VadWithLevel -> VoiceActivityDetectorWrapper 1/2 Internal refactoring of AGC2. This CL is needed in preparation for its child CL to correctly show the upcoming changes in the diff. Bug: webrtc:7494 Change-Id: If7f837e064243d5ffe09e21fc68f489bb00dfdc5 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/234527 Reviewed-by: Hanna Silen <silen@webrtc.org> Commit-Queue: Alessio Bazzica <alessiob@webrtc.org> Cr-Commit-Position: refs/heads/main@{#35170}
2021-10-08 08:15:29 +02:00
":vad_wrapper",
AGC2: adding level estimation option (RMS or peak-based). This CL makes possible to choose the level estimation for the adaptive digital GC of AGC2. The options are RMS (default and currently used estimator) and peak-based (already computed, but not used). Besides adding the new AGC2 config param for the level estimator, this CL also refactors the config class by making it more structured. Bug: webrtc:7494 Change-Id: I20eb558ca50f13536aa7bdea08d21de3b630f8bc Reviewed-on: https://webrtc-review.googlesource.com/c/110144 Commit-Queue: Alessio Bazzica <alessiob@webrtc.org> Reviewed-by: Alex Loiko <aleloi@webrtc.org> Cr-Commit-Position: refs/heads/master@{#25620}
2018-11-13 14:44:15 +01:00
"..: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
"..:apm_logging",
"..:audio_frame_view",
"../../../api:array_view",
"../../../common_audio",
"../../../rtc_base:checks",
AGC2 adaptive digital controller config: new params This CL adds and wires up the following parameters: - VAD probability attack used in `VadLevelAnalyzer` - Adjacent spech frames threshold used in `AdaptiveModeLevelEstimator` - Initial saturation margin used in `AdaptiveModeLevelEstimator` The deprecated ctor in `AdaptiveModeLevelEstimator` is removed. Tested: bit-exactness verified with audioproc_f Bug: webrtc:7494 Change-Id: Idf94aaadba1476757f845e696bfb47ff6252d5f0 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/186048 Commit-Queue: Alessio Bazzica <alessiob@webrtc.org> Reviewed-by: Sam Zackrisson <saza@webrtc.org> Cr-Commit-Position: refs/heads/master@{#32265}
2020-09-30 22:54:00 +02:00
"../../../rtc_base:logging",
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_base:rtc_base_approved",
AGC2 Saturation Protector: switch to ring buffer Even if small, the peak delay buffer copies N-1 elements for each frame whereas a ring buffer is copy-free and scales better if the buffer size increases. Tested: Bit-exactness verified with audioproc_f Bug: webrtc:7494 Change-Id: If8c33877b7ab1d881a0606e222b26857a82fff69 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/184920 Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Commit-Queue: Alessio Bazzica <alessiob@webrtc.org> Cr-Commit-Position: refs/heads/master@{#32165}
2020-09-22 14:44:06 +02:00
"../../../rtc_base:safe_compare",
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_base:safe_minmax",
Remove deprecated field_trial_default and metrics_default. This CL removes some deprecated build targets (and their headers) from system_wrappers: - field_trial_api - field_trial_default - metrics_api - metrics_default It also refreshes all the dependencies on field_trial.h and metrics.h. A nice side effect is that it is finally possible to remove 'nogncheck' from the following files (when it was used with field_trial_default and metrics_default): - sdk/objc/api/peerconnection/RTCMetricsSampleInfo+Private.h - sdk/android/src/jni/pc/peerconnectionfactory.cc - sdk/objc/api/peerconnection/RTCFieldTrials.mm Bug: webrtc:9631 Change-Id: Ib621f41ef8ad0aba4fe1c1d7e749c044afc956c3 No-Try: True Reviewed-on: https://webrtc-review.googlesource.com/100524 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#24878}
2018-09-28 08:51:10 +02:00
"../../../system_wrappers:metrics",
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 Saturation Protector: switch to ring buffer Even if small, the peak delay buffer copies N-1 elements for each frame whereas a ring buffer is copy-free and scales better if the buffer size increases. Tested: Bit-exactness verified with audioproc_f Bug: webrtc:7494 Change-Id: If8c33877b7ab1d881a0606e222b26857a82fff69 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/184920 Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Commit-Queue: Alessio Bazzica <alessiob@webrtc.org> Cr-Commit-Position: refs/heads/master@{#32165}
2020-09-22 14:44:06 +02:00
absl_deps = [ "//third_party/abseil-cpp/absl/types:optional" ]
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
}
Use source_sets in component builds and static_library in release builds. Static libraries don't guarantee that an exported symbol gets linked into a shared library (and in order to support Chromium's component build mode, WebRTC needs to be linked as a shared library). Source sets always pass all the object files to the linker. On the flip side, source_sets link more object files in release builds and to avoid this, this CL introduces a the GN template "rtc_library" that expands to static_library during release builds and to source_set during component builds. See: https://gn.googlesource.com/gn/+/master/docs/reference.md#func_source_set Bug: webrtc:9419 Change-Id: I4667e820c2b3fcec417becbd2034acc13e4f04fe Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/157168 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Reviewed-by: Nico Weber <thakis@chromium.org> Cr-Commit-Position: refs/heads/master@{#29525}
2019-10-17 21:32:04 +02:00
rtc_library("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",
Remove //rtc_base:macromagic from public deps Bug: webrtc:8603 Change-Id: I9708df48c9bde9f86ba2d1a92a278bb0d09f3865 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/257909 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Auto-Submit: Florent Castelli <orphis@webrtc.org> Commit-Queue: Florent Castelli <orphis@webrtc.org> Cr-Commit-Position: refs/heads/main@{#36444}
2022-04-04 15:18:46 +02:00
"../../../rtc_base:macromagic",
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_base:rtc_base_approved",
]
}
AGC2 remove incorrect field trial parsing functions The AGC2 params must be exposed via `AudioProcessing::Config::GainController2` and the Finch params must be parsed in blink (see [1]). Note: this CL breaks the chain of 3 CLs titled "AGC2 AdaptiveModeLevelEstimator min consecutive speech frames". [1] https://source.chromium.org/chromium/chromium/src/+/master:third_party/blink/renderer/modules/mediastream/media_stream_audio_processor.cc;l=593-596?q=HybridAgc&start=11 Bug: webrtc:7494 Change-Id: Ie7bd1bef1d6caf7d2b20600a1626c12171b67c82 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/185044 Commit-Queue: Alessio Bazzica <alessiob@webrtc.org> Reviewed-by: Minyue Li <minyue@webrtc.org> Cr-Commit-Position: refs/heads/master@{#32230}
2020-09-29 14:02:18 +02:00
rtc_source_set("common") {
sources = [ "agc2_common.h" ]
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
}
Use source_sets in component builds and static_library in release builds. Static libraries don't guarantee that an exported symbol gets linked into a shared library (and in order to support Chromium's component build mode, WebRTC needs to be linked as a shared library). Source sets always pass all the object files to the linker. On the flip side, source_sets link more object files in release builds and to avoid this, this CL introduces a the GN template "rtc_library" that expands to static_library during release builds and to source_set during component builds. See: https://gn.googlesource.com/gn/+/master/docs/reference.md#func_source_set Bug: webrtc:9419 Change-Id: I4667e820c2b3fcec417becbd2034acc13e4f04fe Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/157168 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Reviewed-by: Nico Weber <thakis@chromium.org> Cr-Commit-Position: refs/heads/master@{#29525}
2019-10-17 21:32:04 +02:00
rtc_library("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",
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
"interpolated_gain_curve.cc",
"interpolated_gain_curve.h",
AGC2: renaming GainCurveApplier to Limiter. Bug: webrtc:7494 Change-Id: I3dcfb864fd63dbf3f3e7345f8f4cac6c86987e8b Reviewed-on: https://webrtc-review.googlesource.com/c/108581 Reviewed-by: Alex Loiko <aleloi@webrtc.org> Commit-Queue: Alessio Bazzica <alessiob@webrtc.org> Cr-Commit-Position: refs/heads/master@{#25436}
2018-10-30 10:48:38 +01:00
"limiter.cc",
"limiter.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
]
AGC2: `gain_controller2` target isolated Needed to restrict visibility. Bug: webrtc:7494 Change-Id: I58a609666ca04d785c6dd2ed19233b395a94b06c Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/234584 Commit-Queue: Alessio Bazzica <alessiob@webrtc.org> Reviewed-by: Hanna Silen <silen@webrtc.org> Cr-Commit-Position: refs/heads/main@{#35201}
2021-10-14 10:51:40 +02:00
visibility = [
"..:gain_controller2",
"../../audio_mixer:audio_mixer_impl",
"./*",
]
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",
AudioFrameView: size_t -> int Bug: webrtc:7494 Change-Id: I46b1328f3d7da721e144cc3752ed4f458084cf62 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/234522 Reviewed-by: Sam Zackrisson <saza@webrtc.org> Commit-Queue: Alessio Bazzica <alessiob@webrtc.org> Cr-Commit-Position: refs/heads/main@{#35163}
2021-10-07 14:08:59 +02:00
"../../../rtc_base:safe_conversions",
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",
Remove deprecated field_trial_default and metrics_default. This CL removes some deprecated build targets (and their headers) from system_wrappers: - field_trial_api - field_trial_default - metrics_api - metrics_default It also refreshes all the dependencies on field_trial.h and metrics.h. A nice side effect is that it is finally possible to remove 'nogncheck' from the following files (when it was used with field_trial_default and metrics_default): - sdk/objc/api/peerconnection/RTCMetricsSampleInfo+Private.h - sdk/android/src/jni/pc/peerconnectionfactory.cc - sdk/objc/api/peerconnection/RTCFieldTrials.mm Bug: webrtc:9631 Change-Id: Ib621f41ef8ad0aba4fe1c1d7e749c044afc956c3 No-Try: True Reviewed-on: https://webrtc-review.googlesource.com/100524 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#24878}
2018-09-28 08:51:10 +02:00
"../../../system_wrappers:metrics",
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
]
}
Use source_sets in component builds and static_library in release builds. Static libraries don't guarantee that an exported symbol gets linked into a shared library (and in order to support Chromium's component build mode, WebRTC needs to be linked as a shared library). Source sets always pass all the object files to the linker. On the flip side, source_sets link more object files in release builds and to avoid this, this CL introduces a the GN template "rtc_library" that expands to static_library during release builds and to source_set during component builds. See: https://gn.googlesource.com/gn/+/master/docs/reference.md#func_source_set Bug: webrtc:9419 Change-Id: I4667e820c2b3fcec417becbd2034acc13e4f04fe Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/157168 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Reviewed-by: Nico Weber <thakis@chromium.org> Cr-Commit-Position: refs/heads/master@{#29525}
2019-10-17 21:32:04 +02:00
rtc_library("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
]
AGC2: `gain_controller2` target isolated Needed to restrict visibility. Bug: webrtc:7494 Change-Id: I58a609666ca04d785c6dd2ed19233b395a94b06c Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/234584 Commit-Queue: Alessio Bazzica <alessiob@webrtc.org> Reviewed-by: Hanna Silen <silen@webrtc.org> Cr-Commit-Position: refs/heads/main@{#35201}
2021-10-14 10:51:40 +02:00
visibility = [
"..:gain_controller2",
"./*",
]
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",
[Cleanup] Add missing #include. Remove useless ones. This CL is the result of running include-what-you-use tool on part of the code base (audio target and dependencies) plus manual fixes. bug: webrtc:8311 Change-Id: I277d281ce943c3ecc1bd45fd8d83055931743604 Reviewed-on: https://webrtc-review.googlesource.com/c/106280 Commit-Queue: Yves Gerey <yvesg@google.com> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Patrik Höglund <phoglund@webrtc.org> Cr-Commit-Position: refs/heads/master@{#25311}
2018-10-23 12:03:01 +02:00
"../../../api:array_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
"../../../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
]
}
Use source_sets in component builds and static_library in release builds. Static libraries don't guarantee that an exported symbol gets linked into a shared library (and in order to support Chromium's component build mode, WebRTC needs to be linked as a shared library). Source sets always pass all the object files to the linker. On the flip side, source_sets link more object files in release builds and to avoid this, this CL introduces a the GN template "rtc_library" that expands to static_library during release builds and to source_set during component builds. See: https://gn.googlesource.com/gn/+/master/docs/reference.md#func_source_set Bug: webrtc:9419 Change-Id: I4667e820c2b3fcec417becbd2034acc13e4f04fe Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/157168 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Reviewed-by: Nico Weber <thakis@chromium.org> Cr-Commit-Position: refs/heads/master@{#29525}
2019-10-17 21:32:04 +02:00
rtc_library("noise_level_estimator") {
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 = [
"noise_level_estimator.cc",
"noise_level_estimator.h",
]
AGC2: `gain_controller2` target isolated Needed to restrict visibility. Bug: webrtc:7494 Change-Id: I58a609666ca04d785c6dd2ed19233b395a94b06c Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/234584 Commit-Queue: Alessio Bazzica <alessiob@webrtc.org> Reviewed-by: Hanna Silen <silen@webrtc.org> Cr-Commit-Position: refs/heads/main@{#35201}
2021-10-14 10:51:40 +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
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",
"../../../rtc_base:checks",
Reland "Merge cpu_features build targets into //system_wrappers." This is a reland of 2b242d8fba59ccf67e4c5bcf5a1ea80917a23e1c Original change's description: > Merge cpu_features build targets into //system_wrappers. > > Before this CL, functions declared in cpu_features_wrapper.h where > not defined in the same build target, causing brittle builds that > might fail at link time if the binary was not depending on > //system_wrappers (the target with the definitions), violating [1]. > > This CL moves everything into //system_wrappers and also moves > cpu_features_wrapper.h definitions from C to C++ (in order to be able > to add the definitions to a C++ build target like //system_wrappers). > > [1] - https://webrtc.googlesource.com/src/+/refs/heads/master/style-guide.md#h-cc-pairs > > Bug: None > Change-Id: I5a0009cddb17206b19f2a71eeba722faacc4bcae > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/183380 > Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> > Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#32039} TBR=kwiberg@webrtc.org Bug: None Change-Id: I1695b9a34d3ec20c50c1202a745f64fac58edef0 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/183444 Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#32042}
2020-09-05 21:55:35 +02:00
"../../../system_wrappers",
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" ]
}
AGC2: VadWithLevel -> VoiceActivityDetectorWrapper 1/2 Internal refactoring of AGC2. This CL is needed in preparation for its child CL to correctly show the upcoming changes in the diff. Bug: webrtc:7494 Change-Id: If7f837e064243d5ffe09e21fc68f489bb00dfdc5 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/234527 Reviewed-by: Hanna Silen <silen@webrtc.org> Commit-Queue: Alessio Bazzica <alessiob@webrtc.org> Cr-Commit-Position: refs/heads/main@{#35170}
2021-10-08 08:15:29 +02:00
rtc_library("vad_wrapper") {
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
sources = [
AGC2: VadWithLevel -> VoiceActivityDetectorWrapper 1/2 Internal refactoring of AGC2. This CL is needed in preparation for its child CL to correctly show the upcoming changes in the diff. Bug: webrtc:7494 Change-Id: If7f837e064243d5ffe09e21fc68f489bb00dfdc5 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/234527 Reviewed-by: Hanna Silen <silen@webrtc.org> Commit-Queue: Alessio Bazzica <alessiob@webrtc.org> Cr-Commit-Position: refs/heads/main@{#35170}
2021-10-08 08:15:29 +02:00
"vad_wrapper.cc",
"vad_wrapper.h",
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: VAD moved into `GainController2` 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}
2021-10-15 13:57:56 +02:00
visibility = [
"..:gain_controller2",
"./*",
]
RNN VAD optimizations: `VectorMath::DotProduct()` NEON arm64 Results: RNN VAD realtime factor improved from 140x to 195x (+55x) Test device: Pixel 2 XL Benchmark setup: max clock speed forced on all the cores by setting "performance" as scaling governor Bug: webrtc:10480 Change-Id: I3e92f643853ad1fe990db909c578ce78ee826c03 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/198842 Reviewed-by: Per Åhgren <peah@webrtc.org> Commit-Queue: Alessio Bazzica <alessiob@webrtc.org> Cr-Commit-Position: refs/heads/master@{#32888}
2020-12-28 17:57:24 +01:00
defines = []
if (rtc_build_with_neon && current_cpu != "arm64") {
suppressed_configs += [ "//build/config/compiler:compiler_arm_fpu" ]
cflags = [ "-mfpu=neon" ]
}
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
deps = [
AGC2 VAD probability: instant decay / slow attack Feature added to gain robustness to occasional VAD speech probability spikes. In such a case, the attack process reduces the chance that the smoothed values are greater than the speech threshold. Bug: webrtc:7494 Change-Id: I6babe5afe30ea3dea021181a19d86bb74b33a98c Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/185046 Commit-Queue: Alessio Bazzica <alessiob@webrtc.org> Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org> Cr-Commit-Position: refs/heads/master@{#32198}
2020-09-25 14:31:17 +02:00
":common",
AGC2 RNN VAD: safe SIMD optimizations scheme + AVX2 kill switch In preparation for adding AVX2 code, a safe scheme to support different SIMD optimizations is added. Safety features: - AVX2 kill switch to stop using it even if supported by the architecture - struct indicating the available CPU features propagated from AGC2 to each component; in this way - better control over the unit tests - no need to propagate individual kill switches but just set to false features that are turned off Note that (i) this CL does not change the performance of the RNN VAD and (ii) no AVX2 optimization is added yet. Bug: webrtc:10480 Change-Id: I0e61f3311ecd140f38369cf68b6e5954f3dc1f5a Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/193140 Reviewed-by: Per Åhgren <peah@webrtc.org> Commit-Queue: Alessio Bazzica <alessiob@webrtc.org> Cr-Commit-Position: refs/heads/master@{#32739}
2020-11-27 16:02:38 +01:00
":cpu_features",
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
"..: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",
RNN VAD build file refactoring: split into separate (private) targets First step to only expose the relevant RNN VAD API to AGC2. Bug: webrtc:7494 Change-Id: I7f11f6eebded124c30cabd64963c8e3ccc35e58f Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/185124 Commit-Queue: Alessio Bazzica <alessiob@webrtc.org> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#32185}
2020-09-24 15:50:15 +02:00
"rnn_vad:rnn_vad_common",
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 RNN VAD: safe SIMD optimizations scheme + AVX2 kill switch In preparation for adding AVX2 code, a safe scheme to support different SIMD optimizations is added. Safety features: - AVX2 kill switch to stop using it even if supported by the architecture - struct indicating the available CPU features propagated from AGC2 to each component; in this way - better control over the unit tests - no need to propagate individual kill switches but just set to false features that are turned off Note that (i) this CL does not change the performance of the RNN VAD and (ii) no AVX2 optimization is added yet. Bug: webrtc:10480 Change-Id: I0e61f3311ecd140f38369cf68b6e5954f3dc1f5a Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/193140 Reviewed-by: Per Åhgren <peah@webrtc.org> Commit-Queue: Alessio Bazzica <alessiob@webrtc.org> Cr-Commit-Position: refs/heads/master@{#32739}
2020-11-27 16:02:38 +01:00
rtc_library("cpu_features") {
sources = [
"cpu_features.cc",
"cpu_features.h",
]
AGC2: `gain_controller2` target isolated Needed to restrict visibility. Bug: webrtc:7494 Change-Id: I58a609666ca04d785c6dd2ed19233b395a94b06c Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/234584 Commit-Queue: Alessio Bazzica <alessiob@webrtc.org> Reviewed-by: Hanna Silen <silen@webrtc.org> Cr-Commit-Position: refs/heads/main@{#35201}
2021-10-14 10:51:40 +02:00
visibility = [
"..:gain_controller2",
"./*",
]
AGC2 RNN VAD: safe SIMD optimizations scheme + AVX2 kill switch In preparation for adding AVX2 code, a safe scheme to support different SIMD optimizations is added. Safety features: - AVX2 kill switch to stop using it even if supported by the architecture - struct indicating the available CPU features propagated from AGC2 to each component; in this way - better control over the unit tests - no need to propagate individual kill switches but just set to false features that are turned off Note that (i) this CL does not change the performance of the RNN VAD and (ii) no AVX2 optimization is added yet. Bug: webrtc:10480 Change-Id: I0e61f3311ecd140f38369cf68b6e5954f3dc1f5a Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/193140 Reviewed-by: Per Åhgren <peah@webrtc.org> Commit-Queue: Alessio Bazzica <alessiob@webrtc.org> Cr-Commit-Position: refs/heads/master@{#32739}
2020-11-27 16:02:38 +01:00
deps = [
"../../../rtc_base:stringutils",
"../../../rtc_base/system:arch",
"../../../system_wrappers",
]
}
Use source_sets in component builds and static_library in release builds. Static libraries don't guarantee that an exported symbol gets linked into a shared library (and in order to support Chromium's component build mode, WebRTC needs to be linked as a shared library). Source sets always pass all the object files to the linker. On the flip side, source_sets link more object files in release builds and to avoid this, this CL introduces a the GN template "rtc_library" that expands to static_library during release builds and to source_set during component builds. See: https://gn.googlesource.com/gn/+/master/docs/reference.md#func_source_set Bug: webrtc:9419 Change-Id: I4667e820c2b3fcec417becbd2034acc13e4f04fe Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/157168 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Reviewed-by: Nico Weber <thakis@chromium.org> Cr-Commit-Position: refs/heads/master@{#29525}
2019-10-17 21:32:04 +02:00
rtc_library("adaptive_digital_unittests") {
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
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",
AGC2: retuning and large refactoring - Bug fix: the desired initial gain quickly dropped to 0 dB hence starting a call with a too low level - New tuning to make AGC2 more robust to VAD mistakes - Smarter max gain increase speed: to deal with an increased threshold of adjacent speech frames, the gain applier temporarily allows a faster gain increase to deal with a longer time spent waiting for enough speech frames in a row to be observed - Saturation protector isolated from `AdaptiveModeLevelEstimator` to simplify the unit tests for the latter (non bit-exact change) - AGC2 adaptive digital config: unnecessary params deprecated - Code readability improvements - Data dumps clean-up and better naming Bug: webrtc:7494 Change-Id: I4e36059bdf2566cc2a7e1a7e95b7430ba9ae9844 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/215140 Commit-Queue: Alessio Bazzica <alessiob@webrtc.org> Reviewed-by: Jesus de Vicente Pena <devicentepena@webrtc.org> Cr-Commit-Position: refs/heads/master@{#33736}
2021-04-14 19:09:17 +02:00
"saturation_protector_buffer_unittest.cc",
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
"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",
AGC2 config: allow tuning of headroom, max gain and initial gain This CL does *not* change the behavior of the AGC2 adaptive digital controller - bitexactness verified with audioproc_f on a collection of AEC dumps and Wav files (42 recordings in total). Tested: compiled Chrome with this patch and made an appr.tc test call Bug: webrtc:7494 Change-Id: Ia8a9f6fbc3a3459b888a2eed87e108f0d39cfe99 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/233520 Commit-Queue: Alessio Bazzica <alessiob@webrtc.org> Reviewed-by: Sam Zackrisson <saza@webrtc.org> Cr-Commit-Position: refs/heads/main@{#35140}
2021-10-04 13:35:55 +02:00
"..:api",
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
"..: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",
Decouple //rtc_base:rtc_base_tests_utils from gunit. This CL decouples //rtc_base:rtc_base_tests_utils from gunit by moving gunit helpers (rtc_base/gunit.h) and rtc_base/testclient.h (which depends on gunit helpers) to their own build target. It also removes some unused dependencies in the WebRTC build graph. Bug: None Change-Id: Ia9820e84ff697da39b351eef73c45f6e4bdf2623 Reviewed-on: https://webrtc-review.googlesource.com/c/111861 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#25769}
2018-11-23 13:15:08 +01:00
"../../../rtc_base:gunit_helpers",
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_base:rtc_base_approved",
AGC2 remove incorrect field trial parsing functions The AGC2 params must be exposed via `AudioProcessing::Config::GainController2` and the Finch params must be parsed in blink (see [1]). Note: this CL breaks the chain of 3 CLs titled "AGC2 AdaptiveModeLevelEstimator min consecutive speech frames". [1] https://source.chromium.org/chromium/chromium/src/+/master:third_party/blink/renderer/modules/mediastream/media_stream_audio_processor.cc;l=593-596?q=HybridAgc&start=11 Bug: webrtc:7494 Change-Id: Ie7bd1bef1d6caf7d2b20600a1626c12171b67c82 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/185044 Commit-Queue: Alessio Bazzica <alessiob@webrtc.org> Reviewed-by: Minyue Li <minyue@webrtc.org> Cr-Commit-Position: refs/heads/master@{#32230}
2020-09-29 14:02:18 +02:00
"../../../test:test_support",
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
]
}
Use source_sets in component builds and static_library in release builds. Static libraries don't guarantee that an exported symbol gets linked into a shared library (and in order to support Chromium's component build mode, WebRTC needs to be linked as a shared library). Source sets always pass all the object files to the linker. On the flip side, source_sets link more object files in release builds and to avoid this, this CL introduces a the GN template "rtc_library" that expands to static_library during release builds and to source_set during component builds. See: https://gn.googlesource.com/gn/+/master/docs/reference.md#func_source_set Bug: webrtc:9419 Change-Id: I4667e820c2b3fcec417becbd2034acc13e4f04fe Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/157168 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Reviewed-by: Nico Weber <thakis@chromium.org> Cr-Commit-Position: refs/heads/master@{#29525}
2019-10-17 21:32:04 +02:00
rtc_library("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
Reformat GN files. `gn format` recently [1] changed its formatting behavior for deps, source, and a few other elements when they are assigned (with =) single-element lists to be consistent with the formatting of updates (with +=) with single-element. Now that we've rolled in a GN binary with the change, reformat all files so that people don't get presubmit warnings due to this. CL generated with: $ git ls-files | grep BUILD.gn | xargs gn format $ gn format build_overrides/build.gni $ gn format build_overrides/gtest.gni $ gn format modules/audio_coding/audio_coding.gni $ gn format webrtc.gni $ gn format .gn Plus a few manual changes to add exceptions for "public_deps" (after changing these lines the presubmit started to complain). [1] - https://gn-review.googlesource.com/c/gn/+/6860 Bug: webrtc:11302 Change-Id: Iac29d23c1618ebef925c972e2891cd9f4e8cd613 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/166882 Reviewed-by: Patrik Höglund <phoglund@webrtc.org> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Cr-Commit-Position: refs/heads/master@{#30334}
2020-01-21 12:10:10 +01:00
sources = [ "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",
Decouple //rtc_base:rtc_base_tests_utils from gunit. This CL decouples //rtc_base:rtc_base_tests_utils from gunit by moving gunit helpers (rtc_base/gunit.h) and rtc_base/testclient.h (which depends on gunit helpers) to their own build target. It also removes some unused dependencies in the WebRTC build graph. Bug: None Change-Id: Ia9820e84ff697da39b351eef73c45f6e4bdf2623 Reviewed-on: https://webrtc-review.googlesource.com/c/111861 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#25769}
2018-11-23 13:15:08 +01:00
"../../../rtc_base:gunit_helpers",
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
]
}
Use source_sets in component builds and static_library in release builds. Static libraries don't guarantee that an exported symbol gets linked into a shared library (and in order to support Chromium's component build mode, WebRTC needs to be linked as a shared library). Source sets always pass all the object files to the linker. On the flip side, source_sets link more object files in release builds and to avoid this, this CL introduces a the GN template "rtc_library" that expands to static_library during release builds and to source_set during component builds. See: https://gn.googlesource.com/gn/+/master/docs/reference.md#func_source_set Bug: webrtc:9419 Change-Id: I4667e820c2b3fcec417becbd2034acc13e4f04fe Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/157168 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Reviewed-by: Nico Weber <thakis@chromium.org> Cr-Commit-Position: refs/heads/master@{#29525}
2019-10-17 21:32:04 +02:00
rtc_library("fixed_digital_unittests") {
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
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",
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
"interpolated_gain_curve_unittest.cc",
AGC2 Limiter class renamed. Limiter has been renamed to LimiterDbGainCurve, which is a more correct name and will allow in a follow-up CL to reuse the Limiter name for GainCurveApplier. This is done to allow to use the limiter without instancing the fixed digital gain controller and then to fix an AGC2 issue (namely, fixed gain applied after the adaptive one). Bug: webrtc:7494 Change-Id: Icd7050e3e51b832bfbf35e5cc61109215c5b1ca6 Reviewed-on: https://webrtc-review.googlesource.com/c/106901 Commit-Queue: Alessio Bazzica <alessiob@webrtc.org> Reviewed-by: Alex Loiko <aleloi@webrtc.org> Cr-Commit-Position: refs/heads/master@{#25322}
2018-10-18 14:56:06 +02:00
"limiter_db_gain_curve.cc",
"limiter_db_gain_curve.h",
"limiter_db_gain_curve_unittest.cc",
AGC2: renaming GainCurveApplier to Limiter. Bug: webrtc:7494 Change-Id: I3dcfb864fd63dbf3f3e7345f8f4cac6c86987e8b Reviewed-on: https://webrtc-review.googlesource.com/c/108581 Reviewed-by: Alex Loiko <aleloi@webrtc.org> Commit-Queue: Alessio Bazzica <alessiob@webrtc.org> Cr-Commit-Position: refs/heads/master@{#25436}
2018-10-30 10:48:38 +01:00
"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",
Decouple //rtc_base:rtc_base_tests_utils from gunit. This CL decouples //rtc_base:rtc_base_tests_utils from gunit by moving gunit helpers (rtc_base/gunit.h) and rtc_base/testclient.h (which depends on gunit helpers) to their own build target. It also removes some unused dependencies in the WebRTC build graph. Bug: None Change-Id: Ia9820e84ff697da39b351eef73c45f6e4bdf2623 Reviewed-on: https://webrtc-review.googlesource.com/c/111861 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#25769}
2018-11-23 13:15:08 +01:00
"../../../rtc_base:gunit_helpers",
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:rtc_base_approved",
Remove deprecated field_trial_default and metrics_default. This CL removes some deprecated build targets (and their headers) from system_wrappers: - field_trial_api - field_trial_default - metrics_api - metrics_default It also refreshes all the dependencies on field_trial.h and metrics.h. A nice side effect is that it is finally possible to remove 'nogncheck' from the following files (when it was used with field_trial_default and metrics_default): - sdk/objc/api/peerconnection/RTCMetricsSampleInfo+Private.h - sdk/android/src/jni/pc/peerconnectionfactory.cc - sdk/objc/api/peerconnection/RTCFieldTrials.mm Bug: webrtc:9631 Change-Id: Ib621f41ef8ad0aba4fe1c1d7e749c044afc956c3 No-Try: True Reviewed-on: https://webrtc-review.googlesource.com/100524 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#24878}
2018-09-28 08:51:10 +02:00
"../../../system_wrappers:metrics",
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
Use source_sets in component builds and static_library in release builds. Static libraries don't guarantee that an exported symbol gets linked into a shared library (and in order to support Chromium's component build mode, WebRTC needs to be linked as a shared library). Source sets always pass all the object files to the linker. On the flip side, source_sets link more object files in release builds and to avoid this, this CL introduces a the GN template "rtc_library" that expands to static_library during release builds and to source_set during component builds. See: https://gn.googlesource.com/gn/+/master/docs/reference.md#func_source_set Bug: webrtc:9419 Change-Id: I4667e820c2b3fcec417becbd2034acc13e4f04fe Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/157168 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Reviewed-by: Nico Weber <thakis@chromium.org> Cr-Commit-Position: refs/heads/master@{#29525}
2019-10-17 21:32:04 +02:00
rtc_library("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" ]
AGC2: removed unused noise estimator implementation This CL also includes the following changes: - `AudioProcessing::Config::GainController2::noise_estimator` deprecated - `EnergyToDbfs()` optimized by removing unnecessary `sqrt` - Unit test minor fix, incorrect type was used Bug: webrtc:7494 Change-Id: I88a6672d6f7cd03fcf6a3031883522d256880140 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/230940 Reviewed-by: Jesus de Vicente Pena <devicentepena@webrtc.org> Commit-Queue: Alessio Bazzica <alessiob@webrtc.org> Cr-Commit-Position: refs/heads/main@{#34893}
2021-09-01 10:54:47 +02:00
sources = [ "noise_level_estimator_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",
AGC2 noise estimator code style improvements Code style improvements done in preparation for a bug fix (TODO added) which requires changes in the unit tests. Note that one expected value in the unit tests has been adjusted since the white noise generator is now instanced in each separate test and therefore, even if the seed remained the same, the generated sequences differ. Bug: webrtc:7494 Change-Id: I497513b84f50b5c66cf6241a09946ce853eb1cd2 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/214122 Commit-Queue: Alessio Bazzica <alessiob@webrtc.org> Reviewed-by: Ivo Creusen <ivoc@webrtc.org> Cr-Commit-Position: refs/heads/master@{#33636}
2021-04-07 12:03:11 +02:00
"../../../api:function_view",
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
"../../../rtc_base:checks",
Decouple //rtc_base:rtc_base_tests_utils from gunit. This CL decouples //rtc_base:rtc_base_tests_utils from gunit by moving gunit helpers (rtc_base/gunit.h) and rtc_base/testclient.h (which depends on gunit helpers) to their own build target. It also removes some unused dependencies in the WebRTC build graph. Bug: None Change-Id: Ia9820e84ff697da39b351eef73c45f6e4bdf2623 Reviewed-on: https://webrtc-review.googlesource.com/c/111861 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#25769}
2018-11-23 13:15:08 +01:00
"../../../rtc_base:gunit_helpers",
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
"../../../rtc_base:rtc_base_approved",
]
}
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: VadWithLevel -> VoiceActivityDetectorWrapper 1/2 Internal refactoring of AGC2. This CL is needed in preparation for its child CL to correctly show the upcoming changes in the diff. Bug: webrtc:7494 Change-Id: If7f837e064243d5ffe09e21fc68f489bb00dfdc5 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/234527 Reviewed-by: Hanna Silen <silen@webrtc.org> Commit-Queue: Alessio Bazzica <alessiob@webrtc.org> Cr-Commit-Position: refs/heads/main@{#35170}
2021-10-08 08:15:29 +02:00
rtc_library("vad_wrapper_unittests") {
AGC2: adding level estimation option (RMS or peak-based). This CL makes possible to choose the level estimation for the adaptive digital GC of AGC2. The options are RMS (default and currently used estimator) and peak-based (already computed, but not used). Besides adding the new AGC2 config param for the level estimator, this CL also refactors the config class by making it more structured. Bug: webrtc:7494 Change-Id: I20eb558ca50f13536aa7bdea08d21de3b630f8bc Reviewed-on: https://webrtc-review.googlesource.com/c/110144 Commit-Queue: Alessio Bazzica <alessiob@webrtc.org> Reviewed-by: Alex Loiko <aleloi@webrtc.org> Cr-Commit-Position: refs/heads/master@{#25620}
2018-11-13 14:44:15 +01:00
testonly = true
AGC2: VadWithLevel -> VoiceActivityDetectorWrapper 1/2 Internal refactoring of AGC2. This CL is needed in preparation for its child CL to correctly show the upcoming changes in the diff. Bug: webrtc:7494 Change-Id: If7f837e064243d5ffe09e21fc68f489bb00dfdc5 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/234527 Reviewed-by: Hanna Silen <silen@webrtc.org> Commit-Queue: Alessio Bazzica <alessiob@webrtc.org> Cr-Commit-Position: refs/heads/main@{#35170}
2021-10-08 08:15:29 +02:00
sources = [ "vad_wrapper_unittest.cc" ]
AGC2: adding level estimation option (RMS or peak-based). This CL makes possible to choose the level estimation for the adaptive digital GC of AGC2. The options are RMS (default and currently used estimator) and peak-based (already computed, but not used). Besides adding the new AGC2 config param for the level estimator, this CL also refactors the config class by making it more structured. Bug: webrtc:7494 Change-Id: I20eb558ca50f13536aa7bdea08d21de3b630f8bc Reviewed-on: https://webrtc-review.googlesource.com/c/110144 Commit-Queue: Alessio Bazzica <alessiob@webrtc.org> Reviewed-by: Alex Loiko <aleloi@webrtc.org> Cr-Commit-Position: refs/heads/master@{#25620}
2018-11-13 14:44:15 +01:00
deps = [
AGC2 VAD probability: instant decay / slow attack Feature added to gain robustness to occasional VAD speech probability spikes. In such a case, the attack process reduces the chance that the smoothed values are greater than the speech threshold. Bug: webrtc:7494 Change-Id: I6babe5afe30ea3dea021181a19d86bb74b33a98c Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/185046 Commit-Queue: Alessio Bazzica <alessiob@webrtc.org> Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org> Cr-Commit-Position: refs/heads/master@{#32198}
2020-09-25 14:31:17 +02:00
":common",
AGC2: VadWithLevel -> VoiceActivityDetectorWrapper 1/2 Internal refactoring of AGC2. This CL is needed in preparation for its child CL to correctly show the upcoming changes in the diff. Bug: webrtc:7494 Change-Id: If7f837e064243d5ffe09e21fc68f489bb00dfdc5 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/234527 Reviewed-by: Hanna Silen <silen@webrtc.org> Commit-Queue: Alessio Bazzica <alessiob@webrtc.org> Cr-Commit-Position: refs/heads/main@{#35170}
2021-10-08 08:15:29 +02:00
":vad_wrapper",
AGC2: adding level estimation option (RMS or peak-based). This CL makes possible to choose the level estimation for the adaptive digital GC of AGC2. The options are RMS (default and currently used estimator) and peak-based (already computed, but not used). Besides adding the new AGC2 config param for the level estimator, this CL also refactors the config class by making it more structured. Bug: webrtc:7494 Change-Id: I20eb558ca50f13536aa7bdea08d21de3b630f8bc Reviewed-on: https://webrtc-review.googlesource.com/c/110144 Commit-Queue: Alessio Bazzica <alessiob@webrtc.org> Reviewed-by: Alex Loiko <aleloi@webrtc.org> Cr-Commit-Position: refs/heads/master@{#25620}
2018-11-13 14:44:15 +01:00
"..:audio_frame_view",
AGC2: VadWithLevel -> VoiceActivityDetectorWrapper 2/2 Internal refactoring of AGC2 to decouple the VAD, its wrapper and the peak and RMS level measurements. Bit exactness verified with audioproc_f on a collection of AEC dumps and Wav files (42 recordings in total). Bug: webrtc:7494 Change-Id: Ib560f1fcaa601557f4f30e47025c69e91b1b62e0 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/234524 Commit-Queue: Alessio Bazzica <alessiob@webrtc.org> Reviewed-by: Hanna Silen <silen@webrtc.org> Cr-Commit-Position: refs/heads/main@{#35208}
2021-10-14 12:15:20 +02:00
"../../../rtc_base:checks",
Decouple //rtc_base:rtc_base_tests_utils from gunit. This CL decouples //rtc_base:rtc_base_tests_utils from gunit by moving gunit helpers (rtc_base/gunit.h) and rtc_base/testclient.h (which depends on gunit helpers) to their own build target. It also removes some unused dependencies in the WebRTC build graph. Bug: None Change-Id: Ia9820e84ff697da39b351eef73c45f6e4bdf2623 Reviewed-on: https://webrtc-review.googlesource.com/c/111861 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#25769}
2018-11-23 13:15:08 +01:00
"../../../rtc_base:gunit_helpers",
AGC2 VAD probability: instant decay / slow attack Feature added to gain robustness to occasional VAD speech probability spikes. In such a case, the attack process reduces the chance that the smoothed values are greater than the speech threshold. Bug: webrtc:7494 Change-Id: I6babe5afe30ea3dea021181a19d86bb74b33a98c Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/185046 Commit-Queue: Alessio Bazzica <alessiob@webrtc.org> Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org> Cr-Commit-Position: refs/heads/master@{#32198}
2020-09-25 14:31:17 +02:00
"../../../rtc_base:safe_compare",
"../../../test:test_support",
AGC2: adding level estimation option (RMS or peak-based). This CL makes possible to choose the level estimation for the adaptive digital GC of AGC2. The options are RMS (default and currently used estimator) and peak-based (already computed, but not used). Besides adding the new AGC2 config param for the level estimator, this CL also refactors the config class by making it more structured. Bug: webrtc:7494 Change-Id: I20eb558ca50f13536aa7bdea08d21de3b630f8bc Reviewed-on: https://webrtc-review.googlesource.com/c/110144 Commit-Queue: Alessio Bazzica <alessiob@webrtc.org> Reviewed-by: Alex Loiko <aleloi@webrtc.org> Cr-Commit-Position: refs/heads/master@{#25620}
2018-11-13 14:44:15 +01:00
]
}
Use source_sets in component builds and static_library in release builds. Static libraries don't guarantee that an exported symbol gets linked into a shared library (and in order to support Chromium's component build mode, WebRTC needs to be linked as a shared library). Source sets always pass all the object files to the linker. On the flip side, source_sets link more object files in release builds and to avoid this, this CL introduces a the GN template "rtc_library" that expands to static_library during release builds and to source_set during component builds. See: https://gn.googlesource.com/gn/+/master/docs/reference.md#func_source_set Bug: webrtc:9419 Change-Id: I4667e820c2b3fcec417becbd2034acc13e4f04fe Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/157168 Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Reviewed-by: Nico Weber <thakis@chromium.org> Cr-Commit-Position: refs/heads/master@{#29525}
2019-10-17 21:32:04 +02:00
rtc_library("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
Fix AGC2 fixed-adaptive gain controllers order. 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}
2018-11-01 21:31:38 +01:00
visibility = [
":*",
"..:audio_processing_unittests",
]
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",
]
}
Reference in New Issue Copy Permalink