audioproc_f with fake microphone.

webrtc/modules/audio_processing/test/fake_recording_device_unittest.cc

Index: webrtc/modules/audio_processing/test/fake_recording_device_unittest.cc
diff --git a/webrtc/modules/audio_processing/test/fake_recording_device_unittest.cc b/webrtc/modules/audio_processing/test/fake_recording_device_unittest.cc
new file mode 100644
index 0000000000000000000000000000000000000000..e41a1504f450cd8493f79e52069db1a7f6506af2
--- /dev/null
+++ b/webrtc/modules/audio_processing/test/fake_recording_device_unittest.cc
@@ -0,0 +1,30 @@
+/*
+ *  Copyright (c) 2017 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 "webrtc/modules/audio_processing/test/fake_recording_device.h"
+#include "webrtc/test/gtest.h"
+
+TEST(AnalogVolumeMapper, GainCurveShouldBeMonotone) {
+  // using MappingKind = webrtc::FakeRecordingDevice::LevelToScalingMappingKind;
+
+  // for (auto kind : {MappingKind::kLinear}) {
+  //   webrtc::FakeRecordingDevice level_mapper(kind);
+  //   float last_level = level_mapper.GetScalingFactor();
+  //   RTC_DCHECK_LE(0.f, last_level);
+
+  //   for (int i = 0; i < 255; ++i) {
+  //     level_mapper.set_analog_level(i);
+  //     const float current_level = level_mapper.GetScalingFactor();
+  //     RTC_DCHECK_LE(last_level, current_level);
+
+  //     last_level = current_level;
+  //   }
+  // }
+}