Break out Agc code from audio_processing.

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}

modules/audio_processing/BUILD.gn

@@ -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 = [
+  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",
-  ]
 
   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",
     ]
   }

modules/audio_processing/agc/BUILD.gn

@@ -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",
+    ]
+  }
+}

modules/audio_processing/agc/agc_manager_direct.cc

@@ -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"

modules/audio_processing/include/audio_processing.h

@@ -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.

modules/audio_processing/include/gain_control.h

@@ -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_

rtc_tools/BUILD.gn

@@ -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",