audioproc_f with simulated mic analog gain

webrtc/modules/audio_processing/BUILD.gn

Index: webrtc/modules/audio_processing/BUILD.gn
diff --git a/webrtc/modules/audio_processing/BUILD.gn b/webrtc/modules/audio_processing/BUILD.gn
index a569bd100625bfc8e6c8fc44554a5597d08c84d2..2455db41d2ea177e5ee394b6bed4be1251ecfa97 100644
--- a/webrtc/modules/audio_processing/BUILD.gn
+++ b/webrtc/modules/audio_processing/BUILD.gn
@@ -494,6 +494,7 @@ if (rtc_include_tests) {
         ":audioproc_f",
         ":audioproc_unittest_proto",
         ":unpack_aecdump",
+        "aec_dump:aec_dump_unittests",
         "test/conversational_speech",
         "test/py_quality_assessment",
       ]
@@ -524,6 +525,7 @@ if (rtc_include_tests) {
       "config_unittest.cc",
       "echo_cancellation_impl_unittest.cc",
       "splitting_filter_unittest.cc",
+      "test/fake_recording_device_unittest.cc",
       "transient/dyadic_decimator_unittest.cc",
       "transient/file_utils.cc",
       "transient/file_utils.h",
@@ -546,6 +548,7 @@ if (rtc_include_tests) {
     ]
 
     deps = [
+      ":analog_mic_simulation",
       ":audio_processing",
       ":audioproc_test_utils",
       "..:module_api",
@@ -707,6 +710,16 @@ if (rtc_include_tests) {
     }
   }
 
+  rtc_source_set("analog_mic_simulation") {
+    sources = [
+      "test/fake_recording_device.cc",
+      "test/fake_recording_device.h",
+    ]
+    deps = [
+      "../../rtc_base:rtc_base_approved",
+    ]
+  }
+
   if (rtc_enable_protobuf) {
     rtc_executable("unpack_aecdump") {
       testonly = true
@@ -741,6 +754,7 @@ if (rtc_include_tests) {
       ]
 
       deps = [
+        ":analog_mic_simulation",
         ":audio_processing",
         ":audioproc_debug_proto",
         ":audioproc_protobuf_utils",