Remove unused code in APM
- The injection of the AGC2 level estimator into `AgcManagerDirect`
is not used anymore
- `ExperimentalAgc::enabled_agc2_level_estimator` can also be removed
- 3 ctors of `ExperimentalAgc` are unused
- `AgcManagerDirectStandaloneTest::AgcMinMicLevelExperiment` can be
split into separate unit tests (better code clarity)
Bug: webrtc:7494
Change-Id: I5843147c38cf7cb5ee484b0a72fe13dcf363efaf
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/202025
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Reviewed-by: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33027}
diff --git a/modules/audio_processing/agc/BUILD.gn b/modules/audio_processing/agc/BUILD.gn
index 8235456..5ad6644 100644
--- a/modules/audio_processing/agc/BUILD.gn
+++ b/modules/audio_processing/agc/BUILD.gn
@@ -33,7 +33,6 @@
"../../../rtc_base:safe_minmax",
"../../../system_wrappers:field_trial",
"../../../system_wrappers:metrics",
- "../agc2:level_estimation_agc",
"../vad",
]
absl_deps = [ "//third_party/abseil-cpp/absl/types:optional" ]
diff --git a/modules/audio_processing/agc/agc_manager_direct.cc b/modules/audio_processing/agc/agc_manager_direct.cc
index 1428d2a..3f467ce 100644
--- a/modules/audio_processing/agc/agc_manager_direct.cc
+++ b/modules/audio_processing/agc/agc_manager_direct.cc
@@ -16,7 +16,6 @@
#include "common_audio/include/audio_util.h"
#include "modules/audio_processing/agc/gain_control.h"
#include "modules/audio_processing/agc/gain_map_internal.h"
-#include "modules/audio_processing/agc2/adaptive_mode_level_estimator_agc.h"
#include "rtc_base/atomic_ops.h"
#include "rtc_base/checks.h"
#include "rtc_base/logging.h"
@@ -138,24 +137,18 @@
MonoAgc::MonoAgc(ApmDataDumper* data_dumper,
int startup_min_level,
int clipped_level_min,
- bool use_agc2_level_estimation,
bool disable_digital_adaptive,
int min_mic_level)
: min_mic_level_(min_mic_level),
disable_digital_adaptive_(disable_digital_adaptive),
+ agc_(std::make_unique<Agc>()),
max_level_(kMaxMicLevel),
max_compression_gain_(kMaxCompressionGain),
target_compression_(kDefaultCompressionGain),
compression_(target_compression_),
compression_accumulator_(compression_),
startup_min_level_(ClampLevel(startup_min_level, min_mic_level_)),
- clipped_level_min_(clipped_level_min) {
- if (use_agc2_level_estimation) {
- agc_ = std::make_unique<AdaptiveModeLevelEstimatorAgc>(data_dumper);
- } else {
- agc_ = std::make_unique<Agc>();
- }
-}
+ clipped_level_min_(clipped_level_min) {}
MonoAgc::~MonoAgc() = default;
@@ -415,7 +408,6 @@
: AgcManagerDirect(/*num_capture_channels*/ 1,
startup_min_level,
clipped_level_min,
- /*use_agc2_level_estimation*/ false,
/*disable_digital_adaptive*/ false,
sample_rate_hz) {
RTC_DCHECK(channel_agcs_[0]);
@@ -426,7 +418,6 @@
AgcManagerDirect::AgcManagerDirect(int num_capture_channels,
int startup_min_level,
int clipped_level_min,
- bool use_agc2_level_estimation,
bool disable_digital_adaptive,
int sample_rate_hz)
: data_dumper_(
@@ -445,7 +436,7 @@
channel_agcs_[ch] = std::make_unique<MonoAgc>(
data_dumper_ch, startup_min_level, clipped_level_min,
- use_agc2_level_estimation, disable_digital_adaptive_, min_mic_level);
+ disable_digital_adaptive_, min_mic_level);
}
RTC_DCHECK_LT(0, channel_agcs_.size());
channel_agcs_[0]->ActivateLogging();
diff --git a/modules/audio_processing/agc/agc_manager_direct.h b/modules/audio_processing/agc/agc_manager_direct.h
index d3663be..8356b0c 100644
--- a/modules/audio_processing/agc/agc_manager_direct.h
+++ b/modules/audio_processing/agc/agc_manager_direct.h
@@ -38,7 +38,6 @@
AgcManagerDirect(int num_capture_channels,
int startup_min_level,
int clipped_level_min,
- bool use_agc2_level_estimation,
bool disable_digital_adaptive,
int sample_rate_hz);
@@ -73,6 +72,16 @@
DisableDigitalDisablesDigital);
FRIEND_TEST_ALL_PREFIXES(AgcManagerDirectStandaloneTest,
AgcMinMicLevelExperiment);
+ FRIEND_TEST_ALL_PREFIXES(AgcManagerDirectStandaloneTest,
+ AgcMinMicLevelExperimentDisabled);
+ FRIEND_TEST_ALL_PREFIXES(AgcManagerDirectStandaloneTest,
+ AgcMinMicLevelExperimentOutOfRangeAbove);
+ FRIEND_TEST_ALL_PREFIXES(AgcManagerDirectStandaloneTest,
+ AgcMinMicLevelExperimentOutOfRangeBelow);
+ FRIEND_TEST_ALL_PREFIXES(AgcManagerDirectStandaloneTest,
+ AgcMinMicLevelExperimentEnabled50);
+ FRIEND_TEST_ALL_PREFIXES(AgcManagerDirectStandaloneTest,
+ AgcMinMicLevelExperimentEnabledAboveStartupLevel);
// Dependency injection for testing. Don't delete |agc| as the memory is owned
// by the manager.
@@ -106,7 +115,6 @@
MonoAgc(ApmDataDumper* data_dumper,
int startup_min_level,
int clipped_level_min,
- bool use_agc2_level_estimation,
bool disable_digital_adaptive,
int min_mic_level);
~MonoAgc();
diff --git a/modules/audio_processing/agc/agc_manager_direct_unittest.cc b/modules/audio_processing/agc/agc_manager_direct_unittest.cc
index 995801a..c909b08 100644
--- a/modules/audio_processing/agc/agc_manager_direct_unittest.cc
+++ b/modules/audio_processing/agc/agc_manager_direct_unittest.cc
@@ -56,6 +56,13 @@
MOCK_METHOD(bool, stream_is_saturated, (), (const, override));
};
+std::unique_ptr<AgcManagerDirect> CreateAgcManagerDirect(
+ int startup_min_level) {
+ return std::make_unique<AgcManagerDirect>(
+ /*num_capture_channels=*/1, startup_min_level, kClippedMin,
+ /*disable_digital_adaptive=*/true, kSampleRateHz);
+}
+
} // namespace
class AgcManagerDirectTest : public ::testing::Test {
@@ -692,77 +699,78 @@
TEST(AgcManagerDirectStandaloneTest, DisableDigitalDisablesDigital) {
auto agc = std::unique_ptr<Agc>(new ::testing::NiceMock<MockAgc>());
MockGainControl gctrl;
- AgcManagerDirect manager(/* num_capture_channels */ 1, kInitialVolume,
- kClippedMin,
- /* use agc2 level estimation */ false,
- /* disable digital adaptive */ true, kSampleRateHz);
-
EXPECT_CALL(gctrl, set_mode(GainControl::kFixedDigital));
EXPECT_CALL(gctrl, set_target_level_dbfs(0));
EXPECT_CALL(gctrl, set_compression_gain_db(0));
EXPECT_CALL(gctrl, enable_limiter(false));
- manager.Initialize();
- manager.SetupDigitalGainControl(&gctrl);
+ std::unique_ptr<AgcManagerDirect> manager =
+ CreateAgcManagerDirect(kInitialVolume);
+ manager->Initialize();
+ manager->SetupDigitalGainControl(&gctrl);
}
TEST(AgcManagerDirectStandaloneTest, AgcMinMicLevelExperiment) {
- auto agc_man = std::unique_ptr<AgcManagerDirect>(new AgcManagerDirect(
- /* num_capture_channels */ 1, kInitialVolume, kClippedMin, true, true,
- kSampleRateHz));
- EXPECT_EQ(agc_man->channel_agcs_[0]->min_mic_level(), kMinMicLevel);
- EXPECT_EQ(agc_man->channel_agcs_[0]->startup_min_level(), kInitialVolume);
- {
- test::ScopedFieldTrials field_trial(
- "WebRTC-Audio-AgcMinMicLevelExperiment/Disabled/");
- agc_man.reset(new AgcManagerDirect(
- /* num_capture_channels */ 1, kInitialVolume, kClippedMin, true, true,
- kSampleRateHz));
- EXPECT_EQ(agc_man->channel_agcs_[0]->min_mic_level(), kMinMicLevel);
- EXPECT_EQ(agc_man->channel_agcs_[0]->startup_min_level(), kInitialVolume);
- }
- {
- // Valid range of field-trial parameter is [0,255].
- test::ScopedFieldTrials field_trial(
- "WebRTC-Audio-AgcMinMicLevelExperiment/Enabled-256/");
- agc_man.reset(new AgcManagerDirect(
- /* num_capture_channels */ 1, kInitialVolume, kClippedMin, true, true,
- kSampleRateHz));
- EXPECT_EQ(agc_man->channel_agcs_[0]->min_mic_level(), kMinMicLevel);
- EXPECT_EQ(agc_man->channel_agcs_[0]->startup_min_level(), kInitialVolume);
- }
- {
- test::ScopedFieldTrials field_trial(
- "WebRTC-Audio-AgcMinMicLevelExperiment/Enabled--1/");
- agc_man.reset(new AgcManagerDirect(
- /* num_capture_channels */ 1, kInitialVolume, kClippedMin, true, true,
- kSampleRateHz));
- EXPECT_EQ(agc_man->channel_agcs_[0]->min_mic_level(), kMinMicLevel);
- EXPECT_EQ(agc_man->channel_agcs_[0]->startup_min_level(), kInitialVolume);
- }
- {
- // Verify that a valid experiment changes the minimum microphone level.
- // The start volume is larger than the min level and should therefore not
- // be changed.
- test::ScopedFieldTrials field_trial(
- "WebRTC-Audio-AgcMinMicLevelExperiment/Enabled-50/");
- agc_man.reset(new AgcManagerDirect(
- /* num_capture_channels */ 1, kInitialVolume, kClippedMin, true, true,
- kSampleRateHz));
- EXPECT_EQ(agc_man->channel_agcs_[0]->min_mic_level(), 50);
- EXPECT_EQ(agc_man->channel_agcs_[0]->startup_min_level(), kInitialVolume);
- }
- {
- // Use experiment to reduce the default minimum microphone level, start at
- // a lower level and ensure that the startup level is increased to the min
- // level set by the experiment.
- test::ScopedFieldTrials field_trial(
- "WebRTC-Audio-AgcMinMicLevelExperiment/Enabled-50/");
- agc_man.reset(new AgcManagerDirect(/* num_capture_channels */ 1, 30,
- kClippedMin, true, true, kSampleRateHz));
- EXPECT_EQ(agc_man->channel_agcs_[0]->min_mic_level(), 50);
- EXPECT_EQ(agc_man->channel_agcs_[0]->startup_min_level(), 50);
- }
+ std::unique_ptr<AgcManagerDirect> manager =
+ CreateAgcManagerDirect(kInitialVolume);
+ EXPECT_EQ(manager->channel_agcs_[0]->min_mic_level(), kMinMicLevel);
+ EXPECT_EQ(manager->channel_agcs_[0]->startup_min_level(), kInitialVolume);
+}
+
+TEST(AgcManagerDirectStandaloneTest, AgcMinMicLevelExperimentDisabled) {
+ test::ScopedFieldTrials field_trial(
+ "WebRTC-Audio-AgcMinMicLevelExperiment/Disabled/");
+ std::unique_ptr<AgcManagerDirect> manager =
+ CreateAgcManagerDirect(kInitialVolume);
+ EXPECT_EQ(manager->channel_agcs_[0]->min_mic_level(), kMinMicLevel);
+ EXPECT_EQ(manager->channel_agcs_[0]->startup_min_level(), kInitialVolume);
+}
+
+// Checks that a field-trial parameter outside of the valid range [0,255] is
+// ignored.
+TEST(AgcManagerDirectStandaloneTest, AgcMinMicLevelExperimentOutOfRangeAbove) {
+ test::ScopedFieldTrials field_trial(
+ "WebRTC-Audio-AgcMinMicLevelExperiment/Enabled-256/");
+ std::unique_ptr<AgcManagerDirect> manager =
+ CreateAgcManagerDirect(kInitialVolume);
+ EXPECT_EQ(manager->channel_agcs_[0]->min_mic_level(), kMinMicLevel);
+ EXPECT_EQ(manager->channel_agcs_[0]->startup_min_level(), kInitialVolume);
+}
+
+// Checks that a field-trial parameter outside of the valid range [0,255] is
+// ignored.
+TEST(AgcManagerDirectStandaloneTest, AgcMinMicLevelExperimentOutOfRangeBelow) {
+ test::ScopedFieldTrials field_trial(
+ "WebRTC-Audio-AgcMinMicLevelExperiment/Enabled--1/");
+ std::unique_ptr<AgcManagerDirect> manager =
+ CreateAgcManagerDirect(kInitialVolume);
+ EXPECT_EQ(manager->channel_agcs_[0]->min_mic_level(), kMinMicLevel);
+ EXPECT_EQ(manager->channel_agcs_[0]->startup_min_level(), kInitialVolume);
+}
+
+// Verifies that a valid experiment changes the minimum microphone level. The
+// start volume is larger than the min level and should therefore not be
+// changed.
+TEST(AgcManagerDirectStandaloneTest, AgcMinMicLevelExperimentEnabled50) {
+ test::ScopedFieldTrials field_trial(
+ "WebRTC-Audio-AgcMinMicLevelExperiment/Enabled-50/");
+ std::unique_ptr<AgcManagerDirect> manager =
+ CreateAgcManagerDirect(kInitialVolume);
+ EXPECT_EQ(manager->channel_agcs_[0]->min_mic_level(), 50);
+ EXPECT_EQ(manager->channel_agcs_[0]->startup_min_level(), kInitialVolume);
+}
+
+// Uses experiment to reduce the default minimum microphone level, start at a
+// lower level and ensure that the startup level is increased to the min level
+// set by the experiment.
+TEST(AgcManagerDirectStandaloneTest,
+ AgcMinMicLevelExperimentEnabledAboveStartupLevel) {
+ test::ScopedFieldTrials field_trial(
+ "WebRTC-Audio-AgcMinMicLevelExperiment/Enabled-50/");
+ std::unique_ptr<AgcManagerDirect> manager =
+ CreateAgcManagerDirect(/*startup_min_level=*/30);
+ EXPECT_EQ(manager->channel_agcs_[0]->min_mic_level(), 50);
+ EXPECT_EQ(manager->channel_agcs_[0]->startup_min_level(), 50);
}
} // namespace webrtc
diff --git a/modules/audio_processing/agc2/BUILD.gn b/modules/audio_processing/agc2/BUILD.gn
index 7b71f6a..8f2ee0f 100644
--- a/modules/audio_processing/agc2/BUILD.gn
+++ b/modules/audio_processing/agc2/BUILD.gn
@@ -15,31 +15,6 @@
]
}
-rtc_library("level_estimation_agc") {
- sources = [
- "adaptive_mode_level_estimator_agc.cc",
- "adaptive_mode_level_estimator_agc.h",
- ]
- configs += [ "..:apm_debug_dump" ]
- deps = [
- ":adaptive_digital",
- ":common",
- ":gain_applier",
- ":noise_level_estimator",
- ":rnn_vad_with_level",
- "..:api",
- "..:apm_logging",
- "..:audio_frame_view",
- "../../../api:array_view",
- "../../../common_audio",
- "../../../rtc_base:checks",
- "../../../rtc_base:rtc_base_approved",
- "../../../rtc_base:safe_minmax",
- "../agc:level_estimation",
- "../vad",
- ]
-}
-
rtc_library("adaptive_digital") {
sources = [
"adaptive_agc.cc",
diff --git a/modules/audio_processing/agc2/adaptive_mode_level_estimator_agc.cc b/modules/audio_processing/agc2/adaptive_mode_level_estimator_agc.cc
deleted file mode 100644
index 5ceeb7d..0000000
--- a/modules/audio_processing/agc2/adaptive_mode_level_estimator_agc.cc
+++ /dev/null
@@ -1,65 +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/adaptive_mode_level_estimator_agc.h"
-
-#include <cmath>
-#include <vector>
-
-#include "modules/audio_processing/agc2/agc2_common.h"
-#include "modules/audio_processing/include/audio_frame_view.h"
-
-namespace webrtc {
-
-AdaptiveModeLevelEstimatorAgc::AdaptiveModeLevelEstimatorAgc(
- ApmDataDumper* apm_data_dumper)
- : level_estimator_(apm_data_dumper) {
- set_target_level_dbfs(kDefaultAgc2LevelHeadroomDbfs);
-}
-
-// |audio| must be mono; in a multi-channel stream, provide the first (usually
-// left) channel.
-void AdaptiveModeLevelEstimatorAgc::Process(const int16_t* audio,
- size_t length,
- int sample_rate_hz) {
- std::vector<float> float_audio_frame(audio, audio + length);
- const float* const first_channel = &float_audio_frame[0];
- AudioFrameView<const float> frame_view(&first_channel, 1 /* num channels */,
- length);
- const auto vad_prob = agc2_vad_.AnalyzeFrame(frame_view);
- latest_voice_probability_ = vad_prob.speech_probability;
- if (latest_voice_probability_ > kVadConfidenceThreshold) {
- time_in_ms_since_last_estimate_ += kFrameDurationMs;
- }
- level_estimator_.Update(vad_prob);
-}
-
-// Retrieves the difference between the target RMS level and the current
-// signal RMS level in dB. Returns true if an update is available and false
-// otherwise, in which case |error| should be ignored and no action taken.
-bool AdaptiveModeLevelEstimatorAgc::GetRmsErrorDb(int* error) {
- if (time_in_ms_since_last_estimate_ <= kTimeUntilConfidentMs) {
- return false;
- }
- *error =
- std::floor(target_level_dbfs() - level_estimator_.level_dbfs() + 0.5f);
- time_in_ms_since_last_estimate_ = 0;
- return true;
-}
-
-void AdaptiveModeLevelEstimatorAgc::Reset() {
- level_estimator_.Reset();
-}
-
-float AdaptiveModeLevelEstimatorAgc::voice_probability() const {
- return latest_voice_probability_;
-}
-
-} // namespace webrtc
diff --git a/modules/audio_processing/agc2/adaptive_mode_level_estimator_agc.h b/modules/audio_processing/agc2/adaptive_mode_level_estimator_agc.h
deleted file mode 100644
index bc6fa84..0000000
--- a/modules/audio_processing/agc2/adaptive_mode_level_estimator_agc.h
+++ /dev/null
@@ -1,51 +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.
- */
-
-#ifndef MODULES_AUDIO_PROCESSING_AGC2_ADAPTIVE_MODE_LEVEL_ESTIMATOR_AGC_H_
-#define MODULES_AUDIO_PROCESSING_AGC2_ADAPTIVE_MODE_LEVEL_ESTIMATOR_AGC_H_
-
-#include <stddef.h>
-#include <stdint.h>
-
-#include "modules/audio_processing/agc/agc.h"
-#include "modules/audio_processing/agc2/adaptive_mode_level_estimator.h"
-#include "modules/audio_processing/agc2/saturation_protector.h"
-#include "modules/audio_processing/agc2/vad_with_level.h"
-
-namespace webrtc {
-class AdaptiveModeLevelEstimatorAgc : public Agc {
- public:
- explicit AdaptiveModeLevelEstimatorAgc(ApmDataDumper* apm_data_dumper);
-
- // |audio| must be mono; in a multi-channel stream, provide the first (usually
- // left) channel.
- void Process(const int16_t* audio,
- size_t length,
- int sample_rate_hz) override;
-
- // Retrieves the difference between the target RMS level and the current
- // signal RMS level in dB. Returns true if an update is available and false
- // otherwise, in which case |error| should be ignored and no action taken.
- bool GetRmsErrorDb(int* error) override;
- void Reset() override;
-
- float voice_probability() const override;
-
- private:
- static constexpr int kTimeUntilConfidentMs = 700;
- static constexpr int kDefaultAgc2LevelHeadroomDbfs = -1;
- int32_t time_in_ms_since_last_estimate_ = 0;
- AdaptiveModeLevelEstimator level_estimator_;
- VadLevelAnalyzer agc2_vad_;
- float latest_voice_probability_ = 0.f;
-};
-} // namespace webrtc
-
-#endif // MODULES_AUDIO_PROCESSING_AGC2_ADAPTIVE_MODE_LEVEL_ESTIMATOR_AGC_H_
diff --git a/modules/audio_processing/audio_processing_impl.cc b/modules/audio_processing/audio_processing_impl.cc
index 37112f0..f9a6460 100644
--- a/modules/audio_processing/audio_processing_impl.cc
+++ b/modules/audio_processing/audio_processing_impl.cc
@@ -304,8 +304,6 @@
config.Get<ExperimentalAgc>().startup_min_volume;
config_.gain_controller1.analog_gain_controller.clipped_level_min =
config.Get<ExperimentalAgc>().clipped_level_min;
- config_.gain_controller1.analog_gain_controller.enable_agc2_level_estimator =
- config.Get<ExperimentalAgc>().enabled_agc2_level_estimator;
config_.gain_controller1.analog_gain_controller.enable_digital_adaptive =
!config.Get<ExperimentalAgc>().digital_adaptive_disabled;
#endif
@@ -1782,8 +1780,6 @@
num_proc_channels(),
config_.gain_controller1.analog_gain_controller.startup_min_volume,
config_.gain_controller1.analog_gain_controller.clipped_level_min,
- config_.gain_controller1.analog_gain_controller
- .enable_agc2_level_estimator,
!config_.gain_controller1.analog_gain_controller
.enable_digital_adaptive,
capture_nonlocked_.split_rate));
diff --git a/modules/audio_processing/audio_processing_unittest.cc b/modules/audio_processing/audio_processing_unittest.cc
index 545c780..e3aedf6 100644
--- a/modules/audio_processing/audio_processing_unittest.cc
+++ b/modules/audio_processing/audio_processing_unittest.cc
@@ -2931,10 +2931,6 @@
b_analog.clipped_level_min = a_analog.clipped_level_min;
EXPECT_EQ(a, b);
- Toggle(a_analog.enable_agc2_level_estimator);
- b_analog.enable_agc2_level_estimator = a_analog.enable_agc2_level_estimator;
- EXPECT_EQ(a, b);
-
Toggle(a_analog.enable_digital_adaptive);
b_analog.enable_digital_adaptive = a_analog.enable_digital_adaptive;
EXPECT_EQ(a, b);
@@ -2989,10 +2985,6 @@
EXPECT_NE(a, b);
a_analog.clipped_level_min = b_analog.clipped_level_min;
- Toggle(a_analog.enable_agc2_level_estimator);
- EXPECT_NE(a, b);
- a_analog.enable_agc2_level_estimator = b_analog.enable_agc2_level_estimator;
-
Toggle(a_analog.enable_digital_adaptive);
EXPECT_NE(a, b);
a_analog.enable_digital_adaptive = b_analog.enable_digital_adaptive;
diff --git a/modules/audio_processing/include/audio_processing.cc b/modules/audio_processing/include/audio_processing.cc
index 3bc00751..7cb2136 100644
--- a/modules/audio_processing/include/audio_processing.cc
+++ b/modules/audio_processing/include/audio_processing.cc
@@ -87,8 +87,6 @@
analog_lhs.enabled == analog_rhs.enabled &&
analog_lhs.startup_min_volume == analog_rhs.startup_min_volume &&
analog_lhs.clipped_level_min == analog_rhs.clipped_level_min &&
- analog_lhs.enable_agc2_level_estimator ==
- analog_rhs.enable_agc2_level_estimator &&
analog_lhs.enable_digital_adaptive ==
analog_rhs.enable_digital_adaptive;
}
diff --git a/modules/audio_processing/include/audio_processing.h b/modules/audio_processing/include/audio_processing.h
index 942e0c0..1070a56 100644
--- a/modules/audio_processing/include/audio_processing.h
+++ b/modules/audio_processing/include/audio_processing.h
@@ -72,32 +72,13 @@
struct ExperimentalAgc {
ExperimentalAgc() = default;
explicit ExperimentalAgc(bool enabled) : enabled(enabled) {}
- ExperimentalAgc(bool enabled,
- bool enabled_agc2_level_estimator,
- bool digital_adaptive_disabled)
- : enabled(enabled),
- enabled_agc2_level_estimator(enabled_agc2_level_estimator),
- digital_adaptive_disabled(digital_adaptive_disabled) {}
- // Deprecated constructor: will be removed.
- ExperimentalAgc(bool enabled,
- bool enabled_agc2_level_estimator,
- bool digital_adaptive_disabled,
- bool analyze_before_aec)
- : enabled(enabled),
- enabled_agc2_level_estimator(enabled_agc2_level_estimator),
- digital_adaptive_disabled(digital_adaptive_disabled) {}
ExperimentalAgc(bool enabled, int startup_min_volume)
: enabled(enabled), startup_min_volume(startup_min_volume) {}
- ExperimentalAgc(bool enabled, int startup_min_volume, int clipped_level_min)
- : enabled(enabled),
- startup_min_volume(startup_min_volume),
- clipped_level_min(clipped_level_min) {}
static const ConfigOptionID identifier = ConfigOptionID::kExperimentalAgc;
bool enabled = true;
int startup_min_volume = kAgcStartupMinVolume;
// Lowest microphone level that will be applied in response to clipping.
int clipped_level_min = kClippedLevelMin;
- bool enabled_agc2_level_estimator = false;
bool digital_adaptive_disabled = false;
};
@@ -331,7 +312,6 @@
// Lowest analog microphone level that will be applied in response to
// clipping.
int clipped_level_min = kClippedLevelMin;
- bool enable_agc2_level_estimator = false;
bool enable_digital_adaptive = true;
} analog_gain_controller;
} gain_controller1;
diff --git a/modules/audio_processing/test/audio_processing_simulator.cc b/modules/audio_processing/test/audio_processing_simulator.cc
index 40ca7d1..1c8e0a5 100644
--- a/modules/audio_processing/test/audio_processing_simulator.cc
+++ b/modules/audio_processing/test/audio_processing_simulator.cc
@@ -497,11 +497,6 @@
apm_config.gain_controller1.analog_gain_controller.enabled =
*settings_.use_analog_agc;
}
- if (settings_.use_analog_agc_agc2_level_estimator) {
- apm_config.gain_controller1.analog_gain_controller
- .enable_agc2_level_estimator =
- *settings_.use_analog_agc_agc2_level_estimator;
- }
if (settings_.analog_agc_disable_digital_adaptive) {
apm_config.gain_controller1.analog_gain_controller.enable_digital_adaptive =
*settings_.analog_agc_disable_digital_adaptive;
diff --git a/modules/audio_processing/test/audio_processing_simulator.h b/modules/audio_processing/test/audio_processing_simulator.h
index 63e644a..84f8ce4 100644
--- a/modules/audio_processing/test/audio_processing_simulator.h
+++ b/modules/audio_processing/test/audio_processing_simulator.h
@@ -106,7 +106,6 @@
absl::optional<bool> use_vad;
absl::optional<bool> use_le;
absl::optional<bool> use_all;
- absl::optional<bool> use_analog_agc_agc2_level_estimator;
absl::optional<bool> analog_agc_disable_digital_adaptive;
absl::optional<int> agc_mode;
absl::optional<int> agc_target_level;
diff --git a/modules/audio_processing/test/audioproc_float_impl.cc b/modules/audio_processing/test/audioproc_float_impl.cc
index ab395f1..2002e2e 100644
--- a/modules/audio_processing/test/audioproc_float_impl.cc
+++ b/modules/audio_processing/test/audioproc_float_impl.cc
@@ -125,11 +125,6 @@
"Force-deactivate (1) digital adaptation in "
"experimental AGC. Digital adaptation is active by default (0).");
ABSL_FLAG(int,
- analog_agc_agc2_level_estimator,
- kParameterNotSpecifiedValue,
- "AGC2 level estimation"
- " in the experimental AGC. AGC1 level estimation is the default (0)");
-ABSL_FLAG(int,
agc_mode,
kParameterNotSpecifiedValue,
"Specify the AGC mode (0-2)");
@@ -387,8 +382,6 @@
SetSettingIfFlagSet(absl::GetFlag(FLAGS_le), &settings.use_le);
SetSettingIfFlagSet(absl::GetFlag(FLAGS_analog_agc_disable_digital_adaptive),
&settings.analog_agc_disable_digital_adaptive);
- SetSettingIfFlagSet(absl::GetFlag(FLAGS_analog_agc_agc2_level_estimator),
- &settings.use_analog_agc_agc2_level_estimator);
SetSettingIfSpecified(absl::GetFlag(FLAGS_agc_mode), &settings.agc_mode);
SetSettingIfSpecified(absl::GetFlag(FLAGS_agc_target_level),
&settings.agc_target_level);
