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Break out Agc code from audio_processing.

Browse Source 
Splits 'modules/audio_processing:audio_processing' target. The files
in modules/audio_processing/agc now are in targets in that folder.

Reason for doing this was to include
modules/audio_processing/agc/agc.h from another target in the
dependent CL https://webrtc-review.googlesource.com/c/src/+/86603

This could help reducing the binary size in the future.

Bug: webrtc:7494
Change-Id: I61f50ab6d5ce24d19f4097e0f3fa8b0170010887
Reviewed-on: https://webrtc-review.googlesource.com/87422
Commit-Queue: Alex Loiko <aleloi@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23873}
This commit is contained in:
Alex Loiko
2018-07-06 14:54:30 +02:00
committed by Commit Bot
parent 57ec685446
commit ed8ff64ef7
6 changed files with 254 additions and 118 deletions
Download Patch File Download Diff File Expand all files Collapse all files

40
modules/audio_processing/BUILD.gn
View File

@ -34,14 +34,6 @@ rtc_static_library("audio_processing") {
"aec/aec_resampler.h",
"aec/echo_cancellation.cc",
"aec/echo_cancellation.h",
"agc/agc.cc",
"agc/agc.h",
"agc/agc_manager_direct.cc",
"agc/agc_manager_direct.h",
"agc/loudness_histogram.cc",
"agc/loudness_histogram.h",
"agc/utility.cc",
"agc/utility.h",
"audio_buffer.cc",
"audio_buffer.h",
"audio_processing_impl.cc",
@ -108,12 +100,12 @@ rtc_static_library("audio_processing") {
defines = []
deps = [
":aec_core",
":agc_gain_map_internal",
":apm_logging",
":audio_frame_view",
":audio_generator_interface",
":audio_processing_c",
":audio_processing_statistics",
":gain_control_interface",
"../..:typedefs",
"../..:webrtc_common",
"../../api:array_view",
@ -131,6 +123,8 @@ rtc_static_library("audio_processing") {
"../../system_wrappers:cpu_features_api",
"../../system_wrappers:field_trial_api",
"../../system_wrappers:metrics_api",
"agc",
"agc:agc_legacy_c",
"agc2:adaptive_digital",
"agc2:fixed_digital",
"agc2:gain_applier",
@ -169,6 +163,12 @@ rtc_static_library("audio_processing") {
]
}
rtc_source_set("gain_control_interface") {
sources = [
"include/gain_control.h",
]
}
rtc_source_set("audio_processing_statistics") {
visibility = [ "*" ]
sources = [
@ -232,13 +232,7 @@ rtc_source_set("file_audio_generator") {
rtc_source_set("audio_processing_c") {
visibility = [ ":*" ] # Only targets in this file can depend on this.
sources = [
"agc/legacy/analog_agc.c",
"agc/legacy/analog_agc.h",
"agc/legacy/digital_agc.c",
"agc/legacy/digital_agc.h",
"agc/legacy/gain_control.h",
]
sources = []
if (rtc_prefer_fixed_point) {
sources += [
@ -273,6 +267,7 @@ rtc_source_set("audio_processing_c") {
"../../rtc_base:checks",
"../../rtc_base:rtc_base_approved",
"../../system_wrappers:cpu_features_api",
"agc:agc_legacy_c",
]
if (rtc_build_with_neon) {
@ -397,12 +392,6 @@ rtc_source_set("aec_core") {
}
}
rtc_source_set("agc_gain_map_internal") {
sources = [
"agc/gain_map_internal.h",
]
}
if (rtc_include_tests) {
rtc_source_set("mocks") {
testonly = true
@ -446,9 +435,6 @@ if (rtc_include_tests) {
sources = [
"aec/echo_cancellation_unittest.cc",
"aec/system_delay_unittest.cc",
"agc/agc_manager_direct_unittest.cc",
"agc/loudness_histogram_unittest.cc",
"agc/mock_agc.h",
"audio_buffer_unittest.cc",
"audio_frame_view_unittest.cc",
"config_unittest.cc",
@ -496,6 +482,7 @@ if (rtc_include_tests) {
"../../test:test_support",
"../audio_coding:neteq_input_audio_tools",
"aec_dump:mock_aec_dump_unittests",
"agc:agc_unittests",
"agc2:adaptive_digital_unittests",
"agc2:biquad_filter_unittests",
"agc2:fixed_digital_unittests",
@ -620,13 +607,13 @@ if (rtc_include_tests) {
"test/fake_recording_device.h",
]
deps = [
":agc_gain_map_internal",
"../../api:array_view",
"../../api/audio:audio_frame_api",
"../../common_audio:common_audio",
"../../rtc_base:checks",
"../../rtc_base:rtc_base_approved",
"../../rtc_base:safe_minmax",
"agc:gain_map",
"//third_party/abseil-cpp/absl/memory",
"//third_party/abseil-cpp/absl/types:optional",
]
@ -734,6 +721,7 @@ if (rtc_include_tests) {
"../../system_wrappers:metrics_default",
"../../test:fileutils",
"../../test:test_support",
"agc:level_estimation",
"//testing/gtest",
]
}

129
modules/audio_processing/agc/BUILD.gn Normal file
View File

@ -0,0 +1,129 @@
# 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.
import("../../../webrtc.gni")
rtc_source_set("agc") {
sources = [
"agc_manager_direct.cc",
"agc_manager_direct.h",
]
configs += [ "..:apm_debug_dump" ]
deps = [
":gain_map",
":level_estimation",
"..:apm_logging",
"..:gain_control_interface",
"../../..:typedefs",
"../../..:webrtc_common",
"../../../rtc_base:checks",
"../../../rtc_base:logging",
"../../../rtc_base:macromagic",
"../../../rtc_base:safe_minmax",
"../../../system_wrappers:metrics_api",
"../vad",
]
}
rtc_source_set("level_estimation") {
sources = [
"agc.cc",
"agc.h",
"loudness_histogram.cc",
"loudness_histogram.h",
"utility.cc",
"utility.h",
]
deps = [
"../../..:typedefs",
"../../..:webrtc_common",
"../../../rtc_base:checks",
"../../../rtc_base:macromagic",
"../vad",
]
}
rtc_source_set("agc_legacy_c") {
visibility = [
":*",
"..:*",
] # Only targets in this file and in
# audio_processing can depend on
# this.
sources = [
"legacy/analog_agc.c",
"legacy/analog_agc.h",
"legacy/digital_agc.c",
"legacy/digital_agc.h",
"legacy/gain_control.h",
]
deps = [
"../../..:typedefs",
"../../..:webrtc_common",
"../../../common_audio",
"../../../common_audio:common_audio_c",
"../../../common_audio:fft4g",
"../../../rtc_base:checks",
"../../../rtc_base:rtc_base_approved",
"../../../system_wrappers:cpu_features_api",
]
if (rtc_build_with_neon) {
if (current_cpu != "arm64") {
# Enable compilation for the NEON instruction set. This is needed
# since //build/config/arm.gni only enables NEON for iOS, not Android.
# This provides the same functionality as webrtc/build/arm_neon.gypi.
suppressed_configs += [ "//build/config/compiler:compiler_arm_fpu" ]
cflags = [ "-mfpu=neon" ]
}
# Disable LTO on NEON targets due to compiler bug.
# TODO(fdegans): Enable this. See crbug.com/408997.
if (rtc_use_lto) {
cflags -= [
"-flto",
"-ffat-lto-objects",
]
}
}
}
rtc_source_set("gain_map") {
sources = [
"gain_map_internal.h",
]
}
if (rtc_include_tests) {
rtc_source_set("agc_unittests") {
testonly = true
sources = [
"agc_manager_direct_unittest.cc",
"loudness_histogram_unittest.cc",
"mock_agc.h",
]
configs += [ "..:apm_debug_dump" ]
if ((!build_with_chromium || is_win) && is_clang) {
# Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163).
suppressed_configs += [ "//build/config/clang:find_bad_constructs" ]
}
deps = [
":agc",
":level_estimation",
"..:mocks",
"../../..:webrtc_common",
"../../../test:fileutils",
"../../../test:test_support",
"//testing/gtest",
]
}
}

2
modules/audio_processing/agc/agc_manager_direct.cc
View File

@ -17,7 +17,7 @@
#endif
#include "modules/audio_processing/agc/gain_map_internal.h"
#include "modules/audio_processing/gain_control_impl.h"
#include "modules/audio_processing/include/gain_control.h"
#include "rtc_base/checks.h"
#include "rtc_base/logging.h"
#include "rtc_base/numerics/safe_minmax.h"

91
modules/audio_processing/include/audio_processing.h
View File

@ -28,6 +28,7 @@
#include "modules/audio_processing/include/audio_generator.h"
#include "modules/audio_processing/include/audio_processing_statistics.h"
#include "modules/audio_processing/include/config.h"
#include "modules/audio_processing/include/gain_control.h"
#include "rtc_base/arraysize.h"
#include "rtc_base/deprecation.h"
#include "rtc_base/platform_file.h"
@ -948,96 +949,6 @@ class EchoControlMobile {
virtual ~EchoControlMobile() {}
};
// The automatic gain control (AGC) component brings the signal to an
// appropriate range. This is done by applying a digital gain directly and, in
// the analog mode, prescribing an analog gain to be applied at the audio HAL.
//
// Recommended to be enabled on the client-side.
class GainControl {
public:
virtual int Enable(bool enable) = 0;
virtual bool is_enabled() const = 0;
// When an analog mode is set, this must be called prior to |ProcessStream()|
// to pass the current analog level from the audio HAL. Must be within the
// range provided to |set_analog_level_limits()|.
virtual int set_stream_analog_level(int level) = 0;
// When an analog mode is set, this should be called after |ProcessStream()|
// to obtain the recommended new analog level for the audio HAL. It is the
// users responsibility to apply this level.
virtual int stream_analog_level() = 0;
enum Mode {
// Adaptive mode intended for use if an analog volume control is available
// on the capture device. It will require the user to provide coupling
// between the OS mixer controls and AGC through the |stream_analog_level()|
// functions.
//
// It consists of an analog gain prescription for the audio device and a
// digital compression stage.
kAdaptiveAnalog,
// Adaptive mode intended for situations in which an analog volume control
// is unavailable. It operates in a similar fashion to the adaptive analog
// mode, but with scaling instead applied in the digital domain. As with
// the analog mode, it additionally uses a digital compression stage.
kAdaptiveDigital,
// Fixed mode which enables only the digital compression stage also used by
// the two adaptive modes.
//
// It is distinguished from the adaptive modes by considering only a
// short time-window of the input signal. It applies a fixed gain through
// most of the input level range, and compresses (gradually reduces gain
// with increasing level) the input signal at higher levels. This mode is
// preferred on embedded devices where the capture signal level is
// predictable, so that a known gain can be applied.
kFixedDigital
};
virtual int set_mode(Mode mode) = 0;
virtual Mode mode() const = 0;
// Sets the target peak |level| (or envelope) of the AGC in dBFs (decibels
// from digital full-scale). The convention is to use positive values. For
// instance, passing in a value of 3 corresponds to -3 dBFs, or a target
// level 3 dB below full-scale. Limited to [0, 31].
//
// TODO(ajm): use a negative value here instead, if/when VoE will similarly
// update its interface.
virtual int set_target_level_dbfs(int level) = 0;
virtual int target_level_dbfs() const = 0;
// Sets the maximum |gain| the digital compression stage may apply, in dB. A
// higher number corresponds to greater compression, while a value of 0 will
// leave the signal uncompressed. Limited to [0, 90].
virtual int set_compression_gain_db(int gain) = 0;
virtual int compression_gain_db() const = 0;
// When enabled, the compression stage will hard limit the signal to the
// target level. Otherwise, the signal will be compressed but not limited
// above the target level.
virtual int enable_limiter(bool enable) = 0;
virtual bool is_limiter_enabled() const = 0;
// Sets the |minimum| and |maximum| analog levels of the audio capture device.
// Must be set if and only if an analog mode is used. Limited to [0, 65535].
virtual int set_analog_level_limits(int minimum, int maximum) = 0;
virtual int analog_level_minimum() const = 0;
virtual int analog_level_maximum() const = 0;
// Returns true if the AGC has detected a saturation event (period where the
// signal reaches digital full-scale) in the current frame and the analog
// level cannot be reduced.
//
// This could be used as an indicator to reduce or disable analog mic gain at
// the audio HAL.
virtual bool stream_is_saturated() const = 0;
protected:
virtual ~GainControl() {}
};
// TODO(peah): Remove this interface.
// A filtering component which removes DC offset and low-frequency noise.
// Recommended to be enabled on the client-side.

108
modules/audio_processing/include/gain_control.h Normal file
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@ -0,0 +1,108 @@
/*
* 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_INCLUDE_GAIN_CONTROL_H_
#define MODULES_AUDIO_PROCESSING_INCLUDE_GAIN_CONTROL_H_
namespace webrtc {
// The automatic gain control (AGC) component brings the signal to an
// appropriate range. This is done by applying a digital gain directly and, in
// the analog mode, prescribing an analog gain to be applied at the audio HAL.
//
// Recommended to be enabled on the client-side.
class GainControl {
public:
virtual int Enable(bool enable) = 0;
virtual bool is_enabled() const = 0;
// When an analog mode is set, this must be called prior to |ProcessStream()|
// to pass the current analog level from the audio HAL. Must be within the
// range provided to |set_analog_level_limits()|.
virtual int set_stream_analog_level(int level) = 0;
// When an analog mode is set, this should be called after |ProcessStream()|
// to obtain the recommended new analog level for the audio HAL. It is the
// users responsibility to apply this level.
virtual int stream_analog_level() = 0;
enum Mode {
// Adaptive mode intended for use if an analog volume control is available
// on the capture device. It will require the user to provide coupling
// between the OS mixer controls and AGC through the |stream_analog_level()|
// functions.
//
// It consists of an analog gain prescription for the audio device and a
// digital compression stage.
kAdaptiveAnalog,
// Adaptive mode intended for situations in which an analog volume control
// is unavailable. It operates in a similar fashion to the adaptive analog
// mode, but with scaling instead applied in the digital domain. As with
// the analog mode, it additionally uses a digital compression stage.
kAdaptiveDigital,
// Fixed mode which enables only the digital compression stage also used by
// the two adaptive modes.
//
// It is distinguished from the adaptive modes by considering only a
// short time-window of the input signal. It applies a fixed gain through
// most of the input level range, and compresses (gradually reduces gain
// with increasing level) the input signal at higher levels. This mode is
// preferred on embedded devices where the capture signal level is
// predictable, so that a known gain can be applied.
kFixedDigital
};
virtual int set_mode(Mode mode) = 0;
virtual Mode mode() const = 0;
// Sets the target peak |level| (or envelope) of the AGC in dBFs (decibels
// from digital full-scale). The convention is to use positive values. For
// instance, passing in a value of 3 corresponds to -3 dBFs, or a target
// level 3 dB below full-scale. Limited to [0, 31].
//
// TODO(ajm): use a negative value here instead, if/when VoE will similarly
// update its interface.
virtual int set_target_level_dbfs(int level) = 0;
virtual int target_level_dbfs() const = 0;
// Sets the maximum |gain| the digital compression stage may apply, in dB. A
// higher number corresponds to greater compression, while a value of 0 will
// leave the signal uncompressed. Limited to [0, 90].
virtual int set_compression_gain_db(int gain) = 0;
virtual int compression_gain_db() const = 0;
// When enabled, the compression stage will hard limit the signal to the
// target level. Otherwise, the signal will be compressed but not limited
// above the target level.
virtual int enable_limiter(bool enable) = 0;
virtual bool is_limiter_enabled() const = 0;
// Sets the |minimum| and |maximum| analog levels of the audio capture device.
// Must be set if and only if an analog mode is used. Limited to [0, 65535].
virtual int set_analog_level_limits(int minimum, int maximum) = 0;
virtual int analog_level_minimum() const = 0;
virtual int analog_level_maximum() const = 0;
// Returns true if the AGC has detected a saturation event (period where the
// signal reaches digital full-scale) in the current frame and the analog
// level cannot be reduced.
//
// This could be used as an indicator to reduce or disable analog mic gain at
// the audio HAL.
virtual bool stream_is_saturated() const = 0;
protected:
virtual ~GainControl() {}
};
} // namespace webrtc
#endif // MODULES_AUDIO_PROCESSING_INCLUDE_GAIN_CONTROL_H_

2
rtc_tools/BUILD.gn
View File

@ -290,7 +290,7 @@ if (rtc_include_tests) {
deps = [
"../api/audio:audio_frame_api",
"../modules/audio_processing",
"../modules/audio_processing/agc:level_estimation",
"../modules/audio_processing/vad",
"../rtc_base:rtc_base_approved",
"../rtc_base:safe_minmax",