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AGC2 remove incorrect field trial parsing functions

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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}
This commit is contained in:
Alessio Bazzica 2020-09-29 14:02:18 +02:00 committed by Commit Bot
parent 38b768c588
commit 435f279433
6 changed files with 12 additions and 76 deletions
Show Stats Download Patch File Download Diff File Expand all files Collapse all files

13
modules/audio_processing/agc2/BUILD.gn
View File

@ -86,16 +86,8 @@ rtc_library("biquad_filter") {
]
}
rtc_library("common") {
sources = [
"agc2_common.cc",
"agc2_common.h",
]
deps = [
"../../../rtc_base:rtc_base_approved",
"../../../system_wrappers:field_trial",
]
absl_deps = [ "//third_party/abseil-cpp/absl/types:optional" ]
rtc_source_set("common") {
sources = [ "agc2_common.h" ]
}
rtc_library("fixed_digital") {
@ -201,6 +193,7 @@ rtc_library("adaptive_digital_unittests") {
"../../../rtc_base:checks",
"../../../rtc_base:gunit_helpers",
"../../../rtc_base:rtc_base_approved",
"../../../test:test_support",
]
}

8
modules/audio_processing/agc2/adaptive_mode_level_estimator.cc
View File

@ -27,9 +27,9 @@ AdaptiveModeLevelEstimator::AdaptiveModeLevelEstimator(
: AdaptiveModeLevelEstimator(
apm_data_dumper,
AudioProcessing::Config::GainController2::LevelEstimator::kRms,
/*use_saturation_protector=*/true,
GetInitialSaturationMarginDb(),
GetExtraSaturationMarginOffsetDb()) {}
kDefaultUseSaturationProtector,
kDefaultInitialSaturationMarginDb,
kDefaultExtraSaturationMarginDb) {}
AdaptiveModeLevelEstimator::AdaptiveModeLevelEstimator(
ApmDataDumper* apm_data_dumper,
@ -39,7 +39,7 @@ AdaptiveModeLevelEstimator::AdaptiveModeLevelEstimator(
: AdaptiveModeLevelEstimator(apm_data_dumper,
level_estimator,
use_saturation_protector,
GetInitialSaturationMarginDb(),
kDefaultInitialSaturationMarginDb,
extra_saturation_margin_db) {}
AdaptiveModeLevelEstimator::AdaptiveModeLevelEstimator(

3
modules/audio_processing/agc2/adaptive_mode_level_estimator_unittest.cc
View File

@ -46,8 +46,6 @@ struct TestLevelEstimator {
std::unique_ptr<AdaptiveModeLevelEstimator> estimator;
};
} // namespace
TEST(AutomaticGainController2AdaptiveModeLevelEstimator,
EstimatorShouldNotCrash) {
TestLevelEstimator level_estimator;
@ -180,4 +178,5 @@ TEST(AutomaticGainController2AdaptiveModeLevelEstimator,
kMaxDifferenceDb);
}
} // namespace
} // namespace webrtc

58
modules/audio_processing/agc2/agc2_common.cc
View File

@ -1,58 +0,0 @@
/*
* Copyright (c) 2018 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.
*/
#include "modules/audio_processing/agc2/agc2_common.h"
#include <stdio.h>
#include <string>
#include "system_wrappers/include/field_trial.h"
namespace webrtc {
float GetInitialSaturationMarginDb() {
constexpr char kForceInitialSaturationMarginFieldTrial[] =
"WebRTC-Audio-Agc2ForceInitialSaturationMargin";
const bool use_forced_initial_saturation_margin =
webrtc::field_trial::IsEnabled(kForceInitialSaturationMarginFieldTrial);
if (use_forced_initial_saturation_margin) {
const std::string field_trial_string = webrtc::field_trial::FindFullName(
kForceInitialSaturationMarginFieldTrial);
float margin_db = -1;
if (sscanf(field_trial_string.c_str(), "Enabled-%f", &margin_db) == 1 &&
margin_db >= 12.f && margin_db <= 25.f) {
return margin_db;
}
}
constexpr float kDefaultInitialSaturationMarginDb = 20.f;
return kDefaultInitialSaturationMarginDb;
}
float GetExtraSaturationMarginOffsetDb() {
constexpr char kForceExtraSaturationMarginFieldTrial[] =
"WebRTC-Audio-Agc2ForceExtraSaturationMargin";
const bool use_forced_extra_saturation_margin =
webrtc::field_trial::IsEnabled(kForceExtraSaturationMarginFieldTrial);
if (use_forced_extra_saturation_margin) {
const std::string field_trial_string = webrtc::field_trial::FindFullName(
kForceExtraSaturationMarginFieldTrial);
float margin_db = -1;
if (sscanf(field_trial_string.c_str(), "Enabled-%f", &margin_db) == 1 &&
margin_db >= 0.f && margin_db <= 10.f) {
return margin_db;
}
}
constexpr float kDefaultExtraSaturationMarginDb = 2.f;
return kDefaultExtraSaturationMarginDb;
}
} // namespace webrtc

5
modules/audio_processing/agc2/agc2_common.h
View File

@ -53,8 +53,9 @@ constexpr float kInitialSpeechLevelEstimateDbfs = -30.f;
constexpr float kDefaultSmoothedVadProbabilityAttack = 1.f;
// Saturation Protector settings.
float GetInitialSaturationMarginDb();
float GetExtraSaturationMarginOffsetDb();
constexpr bool kDefaultUseSaturationProtector = true;
constexpr float kDefaultInitialSaturationMarginDb = 20.f;
constexpr float kDefaultExtraSaturationMarginDb = 2.f;
constexpr size_t kPeakEnveloperSuperFrameLengthMs = 400;
static_assert(kFullBufferSizeMs % kPeakEnveloperSuperFrameLengthMs == 0,

1
modules/audio_processing/agc2/saturation_protector_unittest.cc
View File

@ -15,6 +15,7 @@
#include "modules/audio_processing/agc2/agc2_common.h"
#include "modules/audio_processing/logging/apm_data_dumper.h"
#include "rtc_base/gunit.h"
#include "test/gmock.h"
namespace webrtc {
namespace {