AGC2: rename AdaptiveDigitalGainApplier -> AdaptiveDigitalGainController
Bug: webrtc:7494
Change-Id: Id45495d1742f7d2027429c97a3b286468da99b1b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/287220
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Reviewed-by: Hanna Silen <silen@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#38857}
diff --git a/modules/audio_processing/BUILD.gn b/modules/audio_processing/BUILD.gn
index 44082f7..79a0255 100644
--- a/modules/audio_processing/BUILD.gn
+++ b/modules/audio_processing/BUILD.gn
@@ -138,7 +138,7 @@
"../../rtc_base:logging",
"../../rtc_base:stringutils",
"../../system_wrappers:field_trial",
- "agc2:adaptive_digital_gain_applier",
+ "agc2:adaptive_digital_gain_controller",
"agc2:cpu_features",
"agc2:fixed_digital",
"agc2:gain_applier",
@@ -420,7 +420,7 @@
"../audio_coding:neteq_input_audio_tools",
"aec_dump:mock_aec_dump_unittests",
"agc:agc_unittests",
- "agc2:adaptive_digital_gain_applier_unittest",
+ "agc2:adaptive_digital_gain_controller_unittest",
"agc2:biquad_filter_unittests",
"agc2:fixed_digital_unittests",
"agc2:gain_applier_unittest",
diff --git a/modules/audio_processing/agc2/BUILD.gn b/modules/audio_processing/agc2/BUILD.gn
index b26d692..220f4d0 100644
--- a/modules/audio_processing/agc2/BUILD.gn
+++ b/modules/audio_processing/agc2/BUILD.gn
@@ -32,10 +32,10 @@
]
}
-rtc_library("adaptive_digital_gain_applier") {
+rtc_library("adaptive_digital_gain_controller") {
sources = [
- "adaptive_digital_gain_applier.cc",
- "adaptive_digital_gain_applier.h",
+ "adaptive_digital_gain_controller.cc",
+ "adaptive_digital_gain_controller.h",
]
visibility = [
@@ -309,14 +309,14 @@
]
}
-rtc_library("adaptive_digital_gain_applier_unittest") {
+rtc_library("adaptive_digital_gain_controller_unittest") {
testonly = true
configs += [ "..:apm_debug_dump" ]
- sources = [ "adaptive_digital_gain_applier_unittest.cc" ]
+ sources = [ "adaptive_digital_gain_controller_unittest.cc" ]
deps = [
- ":adaptive_digital_gain_applier",
+ ":adaptive_digital_gain_controller",
":common",
":test_utils",
"..:api",
diff --git a/modules/audio_processing/agc2/adaptive_digital_gain_applier.cc b/modules/audio_processing/agc2/adaptive_digital_gain_controller.cc
similarity index 96%
rename from modules/audio_processing/agc2/adaptive_digital_gain_applier.cc
rename to modules/audio_processing/agc2/adaptive_digital_gain_controller.cc
index a34f598..b8a99da 100644
--- a/modules/audio_processing/agc2/adaptive_digital_gain_applier.cc
+++ b/modules/audio_processing/agc2/adaptive_digital_gain_controller.cc
@@ -8,7 +8,7 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-#include "modules/audio_processing/agc2/adaptive_digital_gain_applier.h"
+#include "modules/audio_processing/agc2/adaptive_digital_gain_controller.h"
#include <algorithm>
@@ -116,7 +116,7 @@
} // namespace
-AdaptiveDigitalGainApplier::AdaptiveDigitalGainApplier(
+AdaptiveDigitalGainController::AdaptiveDigitalGainController(
ApmDataDumper* apm_data_dumper,
const AudioProcessing::Config::GainController2::AdaptiveDigital& config,
int sample_rate_hz,
@@ -139,8 +139,8 @@
Initialize(sample_rate_hz, num_channels);
}
-void AdaptiveDigitalGainApplier::Initialize(int sample_rate_hz,
- int num_channels) {
+void AdaptiveDigitalGainController::Initialize(int sample_rate_hz,
+ int num_channels) {
if (!config_.dry_run) {
return;
}
@@ -163,8 +163,8 @@
}
}
-void AdaptiveDigitalGainApplier::Process(const FrameInfo& info,
- AudioFrameView<float> frame) {
+void AdaptiveDigitalGainController::Process(const FrameInfo& info,
+ AudioFrameView<float> frame) {
RTC_DCHECK_GE(info.speech_level_dbfs, -150.0f);
RTC_DCHECK_GE(frame.num_channels(), 1);
RTC_DCHECK(
diff --git a/modules/audio_processing/agc2/adaptive_digital_gain_applier.h b/modules/audio_processing/agc2/adaptive_digital_gain_controller.h
similarity index 67%
rename from modules/audio_processing/agc2/adaptive_digital_gain_applier.h
rename to modules/audio_processing/agc2/adaptive_digital_gain_controller.h
index 0b1cceb..05b2ef9 100644
--- a/modules/audio_processing/agc2/adaptive_digital_gain_applier.h
+++ b/modules/audio_processing/agc2/adaptive_digital_gain_controller.h
@@ -8,8 +8,8 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-#ifndef MODULES_AUDIO_PROCESSING_AGC2_ADAPTIVE_DIGITAL_GAIN_APPLIER_H_
-#define MODULES_AUDIO_PROCESSING_AGC2_ADAPTIVE_DIGITAL_GAIN_APPLIER_H_
+#ifndef MODULES_AUDIO_PROCESSING_AGC2_ADAPTIVE_DIGITAL_GAIN_CONTROLLER_H_
+#define MODULES_AUDIO_PROCESSING_AGC2_ADAPTIVE_DIGITAL_GAIN_CONTROLLER_H_
#include <vector>
@@ -21,30 +21,29 @@
class ApmDataDumper;
-// TODO(bugs.webrtc.org/7494): Split into `GainAdaptor` and `GainApplier`.
// Selects the target digital gain, decides when and how quickly to adapt to the
// target and applies the current gain to 10 ms frames.
-class AdaptiveDigitalGainApplier {
+class AdaptiveDigitalGainController {
public:
// Information about a frame to process.
struct FrameInfo {
- float speech_probability; // Probability of speech in the [0, 1] range.
- float speech_level_dbfs; // Estimated speech level (dBFS).
- bool speech_level_reliable; // True with reliable speech level estimation.
- float noise_rms_dbfs; // Estimated noise RMS level (dBFS).
- float headroom_db; // Headroom (dB).
+ float speech_probability; // Probability of speech in the [0, 1] range.
+ float speech_level_dbfs; // Estimated speech level (dBFS).
+ bool speech_level_reliable; // True with reliable speech level estimation.
+ float noise_rms_dbfs; // Estimated noise RMS level (dBFS).
+ float headroom_db; // Headroom (dB).
// TODO(bugs.webrtc.org/7494): Remove `limiter_envelope_dbfs`.
float limiter_envelope_dbfs; // Envelope level from the limiter (dBFS).
};
- AdaptiveDigitalGainApplier(
+ AdaptiveDigitalGainController(
ApmDataDumper* apm_data_dumper,
const AudioProcessing::Config::GainController2::AdaptiveDigital& config,
int sample_rate_hz,
int num_channels);
- AdaptiveDigitalGainApplier(const AdaptiveDigitalGainApplier&) = delete;
- AdaptiveDigitalGainApplier& operator=(const AdaptiveDigitalGainApplier&) =
- delete;
+ AdaptiveDigitalGainController(const AdaptiveDigitalGainController&) = delete;
+ AdaptiveDigitalGainController& operator=(
+ const AdaptiveDigitalGainController&) = delete;
void Initialize(int sample_rate_hz, int num_channels);
@@ -69,4 +68,4 @@
} // namespace webrtc
-#endif // MODULES_AUDIO_PROCESSING_AGC2_ADAPTIVE_DIGITAL_GAIN_APPLIER_H_
+#endif // MODULES_AUDIO_PROCESSING_AGC2_ADAPTIVE_DIGITAL_GAIN_CONTROLLER_H_
diff --git a/modules/audio_processing/agc2/adaptive_digital_gain_applier_unittest.cc b/modules/audio_processing/agc2/adaptive_digital_gain_controller_unittest.cc
similarity index 83%
rename from modules/audio_processing/agc2/adaptive_digital_gain_applier_unittest.cc
rename to modules/audio_processing/agc2/adaptive_digital_gain_controller_unittest.cc
index ea7485f..832be1e 100644
--- a/modules/audio_processing/agc2/adaptive_digital_gain_applier_unittest.cc
+++ b/modules/audio_processing/agc2/adaptive_digital_gain_controller_unittest.cc
@@ -8,7 +8,7 @@
* be found in the AUTHORS file in the root of the source tree.
*/
-#include "modules/audio_processing/agc2/adaptive_digital_gain_applier.h"
+#include "modules/audio_processing/agc2/adaptive_digital_gain_controller.h"
#include <algorithm>
#include <memory>
@@ -48,28 +48,28 @@
constexpr AdaptiveDigitalConfig kDefaultConfig{};
-// Helper to create initialized `AdaptiveDigitalGainApplier` objects.
+// Helper to create initialized `AdaptiveDigitalGainController` objects.
struct GainApplierHelper {
GainApplierHelper(const AdaptiveDigitalConfig& config,
int sample_rate_hz,
int num_channels)
: apm_data_dumper(0),
gain_applier(
- std::make_unique<AdaptiveDigitalGainApplier>(&apm_data_dumper,
- config,
- sample_rate_hz,
- num_channels)) {}
+ std::make_unique<AdaptiveDigitalGainController>(&apm_data_dumper,
+ config,
+ sample_rate_hz,
+ num_channels)) {}
ApmDataDumper apm_data_dumper;
- std::unique_ptr<AdaptiveDigitalGainApplier> gain_applier;
+ std::unique_ptr<AdaptiveDigitalGainController> gain_applier;
};
// Returns a `FrameInfo` sample to simulate noiseless speech detected with
// maximum probability and with level, headroom and limiter envelope chosen
// so that the resulting gain equals the default initial adaptive digital gain
// i.e., no gain adaptation is expected.
-AdaptiveDigitalGainApplier::FrameInfo GetFrameInfoToNotAdapt(
+AdaptiveDigitalGainController::FrameInfo GetFrameInfoToNotAdapt(
const AdaptiveDigitalConfig& config) {
- AdaptiveDigitalGainApplier::FrameInfo info;
+ AdaptiveDigitalGainController::FrameInfo info;
info.speech_probability = kMaxSpeechProbability;
info.speech_level_dbfs = -config.initial_gain_db - config.headroom_db;
info.speech_level_reliable = true;
@@ -79,7 +79,8 @@
return info;
}
-TEST(GainController2AdaptiveGainApplier, GainApplierShouldNotCrash) {
+TEST(GainController2AdaptiveDigitalGainControllerTest,
+ GainApplierShouldNotCrash) {
GainApplierHelper helper(kDefaultConfig, /*sample_rate_hz=*/48000, kStereo);
// Make one call with reasonable audio level values and settings.
VectorFloatFrame fake_audio(kStereo, kFrameLen10ms48kHz, 10000.0f);
@@ -88,7 +89,7 @@
}
// Checks that the maximum allowed gain is applied.
-TEST(GainController2AdaptiveGainApplier, MaxGainApplied) {
+TEST(GainController2AdaptiveDigitalGainControllerTest, MaxGainApplied) {
constexpr int kNumFramesToAdapt =
static_cast<int>(kDefaultConfig.max_gain_db /
GetMaxGainChangePerFrameDb(
@@ -96,7 +97,7 @@
kNumExtraFrames;
GainApplierHelper helper(kDefaultConfig, /*sample_rate_hz=*/8000, kMono);
- AdaptiveDigitalGainApplier::FrameInfo info =
+ AdaptiveDigitalGainController::FrameInfo info =
GetFrameInfoToNotAdapt(kDefaultConfig);
info.speech_level_dbfs = -60.0f;
float applied_gain;
@@ -109,7 +110,7 @@
EXPECT_NEAR(applied_gain_db, kDefaultConfig.max_gain_db, 0.1f);
}
-TEST(GainController2AdaptiveGainApplier, GainDoesNotChangeFast) {
+TEST(GainController2AdaptiveDigitalGainControllerTest, GainDoesNotChangeFast) {
GainApplierHelper helper(kDefaultConfig, /*sample_rate_hz=*/8000, kMono);
constexpr float initial_level_dbfs = -25.0f;
@@ -125,7 +126,7 @@
for (int i = 0; i < kNumFramesToAdapt; ++i) {
SCOPED_TRACE(i);
VectorFloatFrame fake_audio(kMono, kFrameLen10ms8kHz, 1.0f);
- AdaptiveDigitalGainApplier::FrameInfo info =
+ AdaptiveDigitalGainController::FrameInfo info =
GetFrameInfoToNotAdapt(kDefaultConfig);
info.speech_level_dbfs = initial_level_dbfs;
helper.gain_applier->Process(info, fake_audio.float_frame_view());
@@ -139,7 +140,7 @@
for (int i = 0; i < kNumFramesToAdapt; ++i) {
SCOPED_TRACE(i);
VectorFloatFrame fake_audio(kMono, kFrameLen10ms8kHz, 1.0f);
- AdaptiveDigitalGainApplier::FrameInfo info =
+ AdaptiveDigitalGainController::FrameInfo info =
GetFrameInfoToNotAdapt(kDefaultConfig);
info.speech_level_dbfs = 0.f;
helper.gain_applier->Process(info, fake_audio.float_frame_view());
@@ -150,13 +151,13 @@
}
}
-TEST(GainController2AdaptiveGainApplier, GainIsRampedInAFrame) {
+TEST(GainController2AdaptiveDigitalGainControllerTest, GainIsRampedInAFrame) {
GainApplierHelper helper(kDefaultConfig, /*sample_rate_hz=*/48000, kMono);
constexpr float initial_level_dbfs = -25.0f;
VectorFloatFrame fake_audio(kMono, kFrameLen10ms48kHz, 1.0f);
- AdaptiveDigitalGainApplier::FrameInfo info =
+ AdaptiveDigitalGainController::FrameInfo info =
GetFrameInfoToNotAdapt(kDefaultConfig);
info.speech_level_dbfs = initial_level_dbfs;
helper.gain_applier->Process(info, fake_audio.float_frame_view());
@@ -176,7 +177,7 @@
EXPECT_LE(maximal_difference, max_change_per_sample);
}
-TEST(GainController2AdaptiveGainApplier, NoiseLimitsGain) {
+TEST(GainController2AdaptiveDigitalGainControllerTest, NoiseLimitsGain) {
GainApplierHelper helper(kDefaultConfig, /*sample_rate_hz=*/48000, kMono);
constexpr float initial_level_dbfs = -25.0f;
@@ -190,7 +191,7 @@
for (int i = 0; i < num_initial_frames + num_frames; ++i) {
VectorFloatFrame fake_audio(kMono, kFrameLen10ms48kHz, 1.0f);
- AdaptiveDigitalGainApplier::FrameInfo info =
+ AdaptiveDigitalGainController::FrameInfo info =
GetFrameInfoToNotAdapt(kDefaultConfig);
info.speech_level_dbfs = initial_level_dbfs;
info.noise_rms_dbfs = kWithNoiseDbfs;
@@ -207,18 +208,19 @@
}
}
-TEST(GainController2GainApplier, CanHandlePositiveSpeechLevels) {
+TEST(GainController2AdaptiveDigitalGainControllerTest,
+ CanHandlePositiveSpeechLevels) {
GainApplierHelper helper(kDefaultConfig, /*sample_rate_hz=*/48000, kStereo);
// Make one call with positive audio level values and settings.
VectorFloatFrame fake_audio(kStereo, kFrameLen10ms48kHz, 10000.0f);
- AdaptiveDigitalGainApplier::FrameInfo info =
+ AdaptiveDigitalGainController::FrameInfo info =
GetFrameInfoToNotAdapt(kDefaultConfig);
info.speech_level_dbfs = 5.0f;
helper.gain_applier->Process(info, fake_audio.float_frame_view());
}
-TEST(GainController2GainApplier, AudioLevelLimitsGain) {
+TEST(GainController2AdaptiveDigitalGainControllerTest, AudioLevelLimitsGain) {
GainApplierHelper helper(kDefaultConfig, /*sample_rate_hz=*/48000, kMono);
constexpr float initial_level_dbfs = -25.0f;
@@ -232,7 +234,7 @@
for (int i = 0; i < num_initial_frames + num_frames; ++i) {
VectorFloatFrame fake_audio(kMono, kFrameLen10ms48kHz, 1.0f);
- AdaptiveDigitalGainApplier::FrameInfo info =
+ AdaptiveDigitalGainController::FrameInfo info =
GetFrameInfoToNotAdapt(kDefaultConfig);
info.speech_level_dbfs = initial_level_dbfs;
info.limiter_envelope_dbfs = 1.0f;
@@ -250,19 +252,21 @@
}
}
-class AdaptiveDigitalGainApplierTest : public ::testing::TestWithParam<int> {
+class AdaptiveDigitalGainControllerParametrizedTest
+ : public ::testing::TestWithParam<int> {
protected:
int adjacent_speech_frames_threshold() const { return GetParam(); }
};
-TEST_P(AdaptiveDigitalGainApplierTest,
+TEST_P(AdaptiveDigitalGainControllerParametrizedTest,
DoNotIncreaseGainWithTooFewSpeechFrames) {
AdaptiveDigitalConfig config;
config.adjacent_speech_frames_threshold = adjacent_speech_frames_threshold();
GainApplierHelper helper(config, /*sample_rate_hz=*/48000, kMono);
// Lower the speech level so that the target gain will be increased.
- AdaptiveDigitalGainApplier::FrameInfo info = GetFrameInfoToNotAdapt(config);
+ AdaptiveDigitalGainController::FrameInfo info =
+ GetFrameInfoToNotAdapt(config);
info.speech_level_dbfs -= 12.0f;
float prev_gain = 0.0f;
@@ -278,13 +282,15 @@
}
}
-TEST_P(AdaptiveDigitalGainApplierTest, IncreaseGainWithEnoughSpeechFrames) {
+TEST_P(AdaptiveDigitalGainControllerParametrizedTest,
+ IncreaseGainWithEnoughSpeechFrames) {
AdaptiveDigitalConfig config;
config.adjacent_speech_frames_threshold = adjacent_speech_frames_threshold();
GainApplierHelper helper(config, /*sample_rate_hz=*/48000, kMono);
// Lower the speech level so that the target gain will be increased.
- AdaptiveDigitalGainApplier::FrameInfo info = GetFrameInfoToNotAdapt(config);
+ AdaptiveDigitalGainController::FrameInfo info =
+ GetFrameInfoToNotAdapt(config);
info.speech_level_dbfs -= 12.0f;
float prev_gain = 0.0f;
@@ -304,17 +310,19 @@
}
INSTANTIATE_TEST_SUITE_P(GainController2,
- AdaptiveDigitalGainApplierTest,
+ AdaptiveDigitalGainControllerParametrizedTest,
::testing::Values(1, 7, 31));
// Checks that the input is never modified when running in dry run mode.
-TEST(GainController2GainApplier, DryRunDoesNotChangeInput) {
+TEST(GainController2AdaptiveDigitalGainControllerTest,
+ DryRunDoesNotChangeInput) {
AdaptiveDigitalConfig config;
config.dry_run = true;
GainApplierHelper helper(config, /*sample_rate_hz=*/8000, kMono);
// Simulate an input signal with log speech level.
- AdaptiveDigitalGainApplier::FrameInfo info = GetFrameInfoToNotAdapt(config);
+ AdaptiveDigitalGainController::FrameInfo info =
+ GetFrameInfoToNotAdapt(config);
info.speech_level_dbfs = -60.0f;
const int num_frames_to_adapt =
static_cast<int>(
@@ -332,12 +340,14 @@
}
// Checks that no sample is modified before and after the sample rate changes.
-TEST(GainController2GainApplier, DryRunHandlesSampleRateChange) {
+TEST(GainController2AdaptiveDigitalGainControllerTest,
+ DryRunHandlesSampleRateChange) {
AdaptiveDigitalConfig config;
config.dry_run = true;
GainApplierHelper helper(config, /*sample_rate_hz=*/8000, kMono);
- AdaptiveDigitalGainApplier::FrameInfo info = GetFrameInfoToNotAdapt(config);
+ AdaptiveDigitalGainController::FrameInfo info =
+ GetFrameInfoToNotAdapt(config);
info.speech_level_dbfs = -60.0f;
constexpr float kPcmSamples = 123.456f;
VectorFloatFrame fake_audio_8k(kMono, kFrameLen10ms8kHz, kPcmSamples);
@@ -351,12 +361,14 @@
// Checks that no sample is modified before and after the number of channels
// changes.
-TEST(GainController2GainApplier, DryRunHandlesNumChannelsChange) {
+TEST(GainController2AdaptiveDigitalGainControllerTest,
+ DryRunHandlesNumChannelsChange) {
AdaptiveDigitalConfig config;
config.dry_run = true;
GainApplierHelper helper(config, /*sample_rate_hz=*/8000, kMono);
- AdaptiveDigitalGainApplier::FrameInfo info = GetFrameInfoToNotAdapt(config);
+ AdaptiveDigitalGainController::FrameInfo info =
+ GetFrameInfoToNotAdapt(config);
info.speech_level_dbfs = -60.0f;
constexpr float kPcmSamples = 123.456f;
VectorFloatFrame fake_audio_8k(kMono, kFrameLen10ms8kHz, kPcmSamples);
diff --git a/modules/audio_processing/gain_controller2.cc b/modules/audio_processing/gain_controller2.cc
index 6a57dca..d25ce7a 100644
--- a/modules/audio_processing/gain_controller2.cc
+++ b/modules/audio_processing/gain_controller2.cc
@@ -128,8 +128,10 @@
config.adaptive_digital.adjacent_speech_frames_threshold,
&data_dumper_);
// Create controller.
- adaptive_digital_controller_ = std::make_unique<AdaptiveDigitalGainApplier>(
- &data_dumper_, config.adaptive_digital, sample_rate_hz, num_channels);
+ adaptive_digital_controller_ =
+ std::make_unique<AdaptiveDigitalGainController>(
+ &data_dumper_, config.adaptive_digital, sample_rate_hz,
+ num_channels);
}
}
diff --git a/modules/audio_processing/gain_controller2.h b/modules/audio_processing/gain_controller2.h
index fa4743c..7f22f4d 100644
--- a/modules/audio_processing/gain_controller2.h
+++ b/modules/audio_processing/gain_controller2.h
@@ -15,7 +15,7 @@
#include <memory>
#include <string>
-#include "modules/audio_processing/agc2/adaptive_digital_gain_applier.h"
+#include "modules/audio_processing/agc2/adaptive_digital_gain_controller.h"
#include "modules/audio_processing/agc2/cpu_features.h"
#include "modules/audio_processing/agc2/gain_applier.h"
#include "modules/audio_processing/agc2/input_volume_controller.h"
@@ -92,8 +92,7 @@
std::unique_ptr<InputVolumeController> input_volume_controller_;
// TODO(bugs.webrtc.org/7494): Rename to `CrestFactorEstimator`.
std::unique_ptr<SaturationProtector> saturation_protector_;
- // TODO(bugs.webrtc.org/7494): Rename to `AdaptiveDigitalGainController`.
- std::unique_ptr<AdaptiveDigitalGainApplier> adaptive_digital_controller_;
+ std::unique_ptr<AdaptiveDigitalGainController> adaptive_digital_controller_;
Limiter limiter_;
int calls_since_last_limiter_log_;
