audioproc_f with fake microphone.

webrtc/modules/audio_processing/test/fake_recording_device.h

Index: webrtc/modules/audio_processing/test/fake_recording_device.h
diff --git a/webrtc/modules/audio_processing/test/fake_recording_device.h b/webrtc/modules/audio_processing/test/fake_recording_device.h
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+/*
+ *  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.
+ */
+
+#ifndef WEBRTC_MODULES_AUDIO_PROCESSING_TEST_FAKE_RECORDING_DEVICE_H_
+#define WEBRTC_MODULES_AUDIO_PROCESSING_TEST_FAKE_RECORDING_DEVICE_H_
+
+#include <algorithm>
+#include <vector>
+
+#include "webrtc/base/array_view.h"
+#include "webrtc/base/checks.h"
+#include "webrtc/base/optional.h"
+#include "webrtc/modules/include/module_common_types.h"
+
+namespace webrtc {
+
+// Class for simulating an analog gain controller controlled by
+// webrtc::GainControl. The class wraps a mapping from the current
+// level to a floating point scaling factor with the goal of
+// abstracting non-linearities in the gain curve of real analog
+// microphones. The intended mode of operation is
+//
+// set_analog_level(|level suggested by GainControl|);
+// NotifyAudioDeviceLevel(|recorded level of real microphone|); // Optional!
+// ProcessStream(src, dest)
+//
+// In these three calls, the fake device optionally undoes the gain
+// applied by the real microphone. Then in applies a new gain
+// suggested by the AGC.
+class FakeRecordingDevice {
+ public:
+  enum class LevelToScalingMappingKind {
+    kLinear  // A level within [0, 255] is linearly scaled. 0 produces
+             // 0.f, and 255 is 1.0f.
+  };
+
+  explicit FakeRecordingDevice(LevelToScalingMappingKind mapping_kind);
+
+  ~FakeRecordingDevice();
+
+  // Setters/getters for |level_|. The parameter |level| must be
+  // within [0, 255].
+  void set_analog_level(int level);
+  int analog_level() const;
+
+  void ProcessStream(std::vector<rtc::ArrayView<const float>> src,
+                     std::vector<rtc::ArrayView<float>> dest);
+
+  void ProcessStream(const AudioFrame* src, AudioFrame* dest);
+
+  // Meant to be called before ProcessStream if the source signal
+  // passed to ProcessStream has been recorded from a microphone with
+  // known gain.
+  void NotifyAudioDeviceLevel(int level);
+
+ private:
+  // Computes scaling factor based on both the recorded real device
+  // level, gain curve (which depends on |mapping_kind_) and the
+  // current level.
+  float ComputeCompoundScalingFactor();
+
+  // Compute scaling factor based on current gain curve, which depends
+  // on |mapping_kind_|.
+  float GetScalingFactor(int level) const;
+
+  int level_ = 0;
+  rtc::Optional<int> real_device_level_;
+  const LevelToScalingMappingKind mapping_kind_;
+};
+}  // namespace webrtc
+
+#endif  // WEBRTC_MODULES_AUDIO_PROCESSING_TEST_FAKE_RECORDING_DEVICE_H_