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Adaptive Digital gain control structure.

Browse Source 
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}
This commit is contained in:
Alex Loiko 2018-03-27 13:38:36 +02:00 committed by Commit Bot
parent 79aab3ff51
commit 2bac896d5e
13 changed files with 412 additions and 9 deletions
Show Stats Download Patch File Download Diff File Expand all files Collapse all files

2
modules/audio_mixer/BUILD.gn
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@ -51,7 +51,7 @@ rtc_static_library("audio_mixer_impl") {
"../audio_processing",
"../audio_processing:apm_logging",
"../audio_processing:audio_frame_view",
"../audio_processing/agc2:agc2",
"../audio_processing/agc2:fixed_digital",
]
}

3
modules/audio_processing/BUILD.gn
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@ -141,7 +141,8 @@ rtc_static_library("audio_processing") {
"../../system_wrappers:cpu_features_api",
"../../system_wrappers:field_trial_api",
"../../system_wrappers:metrics_api",
"agc2",
"agc2:adaptive_digital",
"agc2:fixed_digital",
"vad",
]

73
modules/audio_processing/agc2/BUILD.gn
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@ -8,9 +8,44 @@
import("../../../webrtc.gni")
rtc_source_set("agc2") {
group("agc2") {
deps = [
":adaptive_digital",
":fixed_digital",
]
}
rtc_source_set("adaptive_digital") {
sources = [
"adaptive_agc.cc",
"adaptive_agc.h",
"adaptive_digital_gain_applier.cc",
"adaptive_digital_gain_applier.h",
"adaptive_mode_level_estimator.cc",
"adaptive_mode_level_estimator.h",
"noise_level_estimator.cc",
"noise_level_estimator.h",
]
configs += [ "..:apm_debug_dump" ]
deps = [
":common",
"..:aec_core",
"..:apm_logging",
"..:audio_frame_view",
"../../../api:array_view",
"../../../common_audio",
"../../../rtc_base:checks",
"../../../rtc_base:rtc_base_approved",
"../../../rtc_base:safe_minmax",
"../vad",
"../vad:vad_with_level",
]
}
rtc_source_set("fixed_digital") {
sources = [
"agc2_common.h",
"fixed_digital_level_estimator.cc",
"fixed_digital_level_estimator.h",
"fixed_gain_controller.cc",
@ -24,6 +59,7 @@ rtc_source_set("agc2") {
configs += [ "..:apm_debug_dump" ]
deps = [
":common",
"..:apm_logging",
"..:audio_frame_view",
"../../../api:array_view",
@ -36,13 +72,36 @@ rtc_source_set("agc2") {
]
}
rtc_source_set("common") {
sources = [
"agc2_common.h",
]
deps = [
"../../../rtc_base:rtc_base_approved",
]
}
rtc_source_set("test_utils") {
testonly = true
visibility = [ ":*" ]
sources = [
"agc2_testing_common.cc",
"agc2_testing_common.h",
"vector_float_frame.cc",
"vector_float_frame.h",
]
deps = [
"..:audio_frame_view",
"../../../rtc_base:checks",
"../../../rtc_base:rtc_base_approved",
]
}
rtc_source_set("fixed_digital_unittests") {
testonly = true
configs += [ "..:apm_debug_dump" ]
sources = [
"agc2_testing_common.cc",
"agc2_testing_common.h",
"agc2_testing_common_unittest.cc",
"compute_interpolated_gain_curve.cc",
"compute_interpolated_gain_curve.h",
@ -53,11 +112,11 @@ rtc_source_set("fixed_digital_unittests") {
"limiter.cc",
"limiter.h",
"limiter_unittest.cc",
"vector_float_frame.cc",
"vector_float_frame.h",
]
deps = [
":agc2",
":common",
":fixed_digital",
":test_utils",
"..:apm_logging",
"..:audio_frame_view",
"../../../api:array_view",

59
modules/audio_processing/agc2/adaptive_agc.cc Normal file
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@ -0,0 +1,59 @@
/*
* 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_agc.h"
#include <algorithm>
#include <numeric>
#include "common_audio/include/audio_util.h"
#include "modules/audio_processing/logging/apm_data_dumper.h"
#include "modules/audio_processing/vad/voice_activity_detector.h"
namespace webrtc {
AdaptiveAgc::AdaptiveAgc(ApmDataDumper* apm_data_dumper)
: speech_level_estimator_(apm_data_dumper),
gain_applier_(apm_data_dumper),
apm_data_dumper_(apm_data_dumper) {
RTC_DCHECK(apm_data_dumper);
}
AdaptiveAgc::~AdaptiveAgc() = default;
void AdaptiveAgc::Process(AudioFrameView<float> float_frame) {
// Some VADs are 'bursty'. They return several estimates for some
// frames, and no estimates for other frames. We want to feed all to
// the level estimator, but only care about the last level it
// produces.
rtc::ArrayView<const VadWithLevel::LevelAndProbability> vad_results =
vad_.AnalyzeFrame(float_frame);
for (const auto& vad_result : vad_results) {
apm_data_dumper_->DumpRaw("agc2_vad_probability",
vad_result.speech_probability);
apm_data_dumper_->DumpRaw("agc2_vad_rms_dbfs", vad_result.speech_rms_dbfs);
apm_data_dumper_->DumpRaw("agc2_vad_peak_dbfs",
vad_result.speech_peak_dbfs);
speech_level_estimator_.UpdateEstimation(vad_result);
}
const float speech_level_dbfs = speech_level_estimator_.LatestLevelEstimate();
const float noise_level_dbfs = noise_level_estimator_.Analyze(float_frame);
apm_data_dumper_->DumpRaw("agc2_noise_estimate_dbfs", noise_level_dbfs);
// The gain applier applies the gain.
gain_applier_.Process(speech_level_dbfs, noise_level_dbfs, vad_results,
float_frame);
}
} // namespace webrtc

41
modules/audio_processing/agc2/adaptive_agc.h Normal file
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@ -0,0 +1,41 @@
/*
* 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_AGC_H_
#define MODULES_AUDIO_PROCESSING_AGC2_ADAPTIVE_AGC_H_
#include <memory>
#include "modules/audio_processing/agc2/adaptive_digital_gain_applier.h"
#include "modules/audio_processing/agc2/adaptive_mode_level_estimator.h"
#include "modules/audio_processing/agc2/noise_level_estimator.h"
#include "modules/audio_processing/include/audio_frame_view.h"
#include "modules/audio_processing/vad/vad_with_level.h"
namespace webrtc {
class ApmDataDumper;
class AdaptiveAgc {
public:
explicit AdaptiveAgc(ApmDataDumper* apm_data_dumper);
void Process(AudioFrameView<float> float_frame);
~AdaptiveAgc();
private:
AdaptiveModeLevelEstimator speech_level_estimator_;
VadWithLevel vad_;
AdaptiveDigitalGainApplier gain_applier_;
ApmDataDumper* const apm_data_dumper_;
NoiseLevelEstimator noise_level_estimator_;
};
} // namespace webrtc
#endif // MODULES_AUDIO_PROCESSING_AGC2_ADAPTIVE_AGC_H_

40
modules/audio_processing/agc2/adaptive_digital_gain_applier.cc Normal file
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@ -0,0 +1,40 @@
/*
* 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_digital_gain_applier.h"
#include <algorithm>
#include "common_audio/include/audio_util.h"
#include "modules/audio_processing/agc2/agc2_common.h"
#include "modules/audio_processing/logging/apm_data_dumper.h"
namespace webrtc {
AdaptiveDigitalGainApplier::AdaptiveDigitalGainApplier(
ApmDataDumper* apm_data_dumper)
: apm_data_dumper_(apm_data_dumper) {}
void AdaptiveDigitalGainApplier::Process(
float input_level_dbfs,
float input_noise_level_dbfs,
rtc::ArrayView<const VadWithLevel::LevelAndProbability> vad_results,
AudioFrameView<float> float_frame) {
RTC_DCHECK_GE(input_level_dbfs, -150.f);
RTC_DCHECK_LE(input_level_dbfs, 0.f);
RTC_DCHECK_GE(float_frame.num_channels(), 1);
RTC_DCHECK_GE(float_frame.samples_per_channel(), 1);
// TODO(webrtc:8925): compute and apply the gain.
last_gain_db_ = 1.f;
apm_data_dumper_->DumpRaw("agc2_applied_gain_db", last_gain_db_);
}
} // namespace webrtc

37
modules/audio_processing/agc2/adaptive_digital_gain_applier.h Normal file
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@ -0,0 +1,37 @@
/*
* 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_DIGITAL_GAIN_APPLIER_H_
#define MODULES_AUDIO_PROCESSING_AGC2_ADAPTIVE_DIGITAL_GAIN_APPLIER_H_
#include "modules/audio_processing/include/audio_frame_view.h"
#include "modules/audio_processing/vad/vad_with_level.h"
namespace webrtc {
class ApmDataDumper;
class AdaptiveDigitalGainApplier {
public:
explicit AdaptiveDigitalGainApplier(ApmDataDumper* apm_data_dumper);
// Decide what gain to apply.
void Process(
float input_level_dbfs,
float input_noise_level_dbfs,
rtc::ArrayView<const VadWithLevel::LevelAndProbability> vad_results,
AudioFrameView<float> float_frame);
private:
float last_gain_db_ = 0.f;
ApmDataDumper* apm_data_dumper_ = nullptr;
};
} // namespace webrtc
#endif // MODULES_AUDIO_PROCESSING_AGC2_ADAPTIVE_DIGITAL_GAIN_APPLIER_H_

36
modules/audio_processing/agc2/adaptive_mode_level_estimator.cc Normal file
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@ -0,0 +1,36 @@
/*
* 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.h"
#include "modules/audio_processing/agc2/agc2_common.h"
#include "modules/audio_processing/logging/apm_data_dumper.h"
#include "rtc_base/numerics/safe_minmax.h"
namespace webrtc {
AdaptiveModeLevelEstimator::AdaptiveModeLevelEstimator(
ApmDataDumper* apm_data_dumper) {}
void AdaptiveModeLevelEstimator::UpdateEstimation(
const VadWithLevel::LevelAndProbability& vad_data) {
RTC_DCHECK_GT(vad_data.speech_rms_dbfs, -150.f);
RTC_DCHECK_LT(vad_data.speech_rms_dbfs, 50.f);
RTC_DCHECK_GT(vad_data.speech_peak_dbfs, -150.f);
RTC_DCHECK_LT(vad_data.speech_peak_dbfs, 50.f);
RTC_DCHECK_GE(vad_data.speech_probability, 0.f);
RTC_DCHECK_LE(vad_data.speech_probability, 1.f);
}
float AdaptiveModeLevelEstimator::LatestLevelEstimate() const {
// TODO(webrtc:7494): This is a stub. Add implementation.
return 0.f;
}
} // namespace webrtc

28
modules/audio_processing/agc2/adaptive_mode_level_estimator.h Normal file
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@ -0,0 +1,28 @@
/*
* 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_H_
#define MODULES_AUDIO_PROCESSING_AGC2_ADAPTIVE_MODE_LEVEL_ESTIMATOR_H_
#include "modules/audio_processing/vad/vad_with_level.h"
namespace webrtc {
class ApmDataDumper;
class AdaptiveModeLevelEstimator {
public:
explicit AdaptiveModeLevelEstimator(ApmDataDumper* apm_data_dumper);
void UpdateEstimation(const VadWithLevel::LevelAndProbability& vad_data);
float LatestLevelEstimate() const;
};
} // namespace webrtc
#endif // MODULES_AUDIO_PROCESSING_AGC2_ADAPTIVE_MODE_LEVEL_ESTIMATOR_H_

20
modules/audio_processing/agc2/noise_level_estimator.cc Normal file
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@ -0,0 +1,20 @@
/*
* Copyright (c) 2016 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/noise_level_estimator.h"
namespace webrtc {
float NoiseLevelEstimator::Analyze(AudioFrameView<const float> frame) {
// TODO(webrtc:7494): This is a stub. Add implementation.
return -50.f;
}
} // namespace webrtc

32
modules/audio_processing/agc2/noise_level_estimator.h Normal file
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@ -0,0 +1,32 @@
/*
* Copyright (c) 2016 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_NOISE_LEVEL_ESTIMATOR_H_
#define MODULES_AUDIO_PROCESSING_AGC2_NOISE_LEVEL_ESTIMATOR_H_
#include "modules/audio_processing/include/audio_frame_view.h"
#include "rtc_base/constructormagic.h"
namespace webrtc {
class NoiseLevelEstimator {
public:
NoiseLevelEstimator() {}
// Returns the estimated noise level in DbFS.
float Analyze(AudioFrameView<const float> frame);
private:
RTC_DISALLOW_COPY_AND_ASSIGN(NoiseLevelEstimator);
};
} // namespace webrtc
#endif // MODULES_AUDIO_PROCESSING_AGC2_NOISE_LEVEL_ESTIMATOR_H_

10
modules/audio_processing/vad/BUILD.gn
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@ -45,6 +45,16 @@ rtc_static_library("vad") {
]
}
rtc_source_set("vad_with_level") {
sources = [
"vad_with_level.h",
]
deps = [
"..:audio_frame_view",
"../../../api:array_view",
]
}
if (rtc_include_tests) {
rtc_static_library("vad_unittests") {
testonly = true

40
modules/audio_processing/vad/vad_with_level.h Normal file
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@ -0,0 +1,40 @@
/*
* 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_VAD_VAD_WITH_LEVEL_H_
#define MODULES_AUDIO_PROCESSING_VAD_VAD_WITH_LEVEL_H_
#include "api/array_view.h"
#include "modules/audio_processing/include/audio_frame_view.h"
namespace webrtc {
class VadWithLevel {
public:
struct LevelAndProbability {
constexpr LevelAndProbability(float prob, float rms, float peak)
: speech_probability(prob),
speech_rms_dbfs(rms),
speech_peak_dbfs(peak) {}
LevelAndProbability() = default;
float speech_probability = 0;
float speech_rms_dbfs = 0; // Root mean square in decibels to full-scale.
float speech_peak_dbfs = 0;
};
// TODO(webrtc:7494): This is a stub. Add implementation.
rtc::ArrayView<const LevelAndProbability> AnalyzeFrame(
AudioFrameView<const float> frame) {
return {nullptr, 0};
}
};
} // namespace webrtc
#endif // MODULES_AUDIO_PROCESSING_VAD_VAD_WITH_LEVEL_H_