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}
This commit is contained in:
Alessio Bazzica 2022-12-09 08:46:06 +01:00 committed by WebRTC LUCI CQ
parent c1080dc884
commit f72bc5f1e2
7 changed files with 80 additions and 68 deletions

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@ -138,7 +138,7 @@ rtc_library("gain_controller2") {
"../../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 @@ if (rtc_include_tests) {
"../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",

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@ -32,10 +32,10 @@ rtc_library("speech_level_estimator") {
]
}
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("speech_level_estimator_unittest") {
]
}
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",

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@ -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 @@ void CopyAudio(AudioFrameView<const float> src,
} // namespace
AdaptiveDigitalGainApplier::AdaptiveDigitalGainApplier(
AdaptiveDigitalGainController::AdaptiveDigitalGainController(
ApmDataDumper* apm_data_dumper,
const AudioProcessing::Config::GainController2::AdaptiveDigital& config,
int sample_rate_hz,
@ -139,8 +139,8 @@ AdaptiveDigitalGainApplier::AdaptiveDigitalGainApplier(
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::Initialize(int sample_rate_hz,
}
}
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(

View File

@ -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 @@ namespace webrtc {
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 @@ class AdaptiveDigitalGainApplier {
} // namespace webrtc
#endif // MODULES_AUDIO_PROCESSING_AGC2_ADAPTIVE_DIGITAL_GAIN_APPLIER_H_
#endif // MODULES_AUDIO_PROCESSING_AGC2_ADAPTIVE_DIGITAL_GAIN_CONTROLLER_H_

View File

@ -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 @@ using AdaptiveDigitalConfig =
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 @@ AdaptiveDigitalGainApplier::FrameInfo GetFrameInfoToNotAdapt(
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 @@ TEST(GainController2AdaptiveGainApplier, GainApplierShouldNotCrash) {
}
// 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 @@ TEST(GainController2AdaptiveGainApplier, MaxGainApplied) {
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 @@ TEST(GainController2AdaptiveGainApplier, MaxGainApplied) {
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 @@ TEST(GainController2AdaptiveGainApplier, GainDoesNotChangeFast) {
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 @@ TEST(GainController2AdaptiveGainApplier, GainDoesNotChangeFast) {
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, GainDoesNotChangeFast) {
}
}
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 @@ TEST(GainController2AdaptiveGainApplier, GainIsRampedInAFrame) {
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 @@ TEST(GainController2AdaptiveGainApplier, NoiseLimitsGain) {
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(GainController2AdaptiveGainApplier, NoiseLimitsGain) {
}
}
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 @@ TEST(GainController2GainApplier, AudioLevelLimitsGain) {
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 @@ TEST(GainController2GainApplier, AudioLevelLimitsGain) {
}
}
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,
}
}
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 @@ TEST_P(AdaptiveDigitalGainApplierTest, IncreaseGainWithEnoughSpeechFrames) {
}
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 @@ TEST(GainController2GainApplier, DryRunDoesNotChangeInput) {
}
// 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 @@ TEST(GainController2GainApplier, DryRunHandlesSampleRateChange) {
// 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);

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@ -128,8 +128,10 @@ GainController2::GainController2(
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);
}
}

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@ -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 @@ class GainController2 {
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_;