audioproc_f with simulated mic analog gain

webrtc/modules/audio_processing/test/fake_recording_device.h

+/*
+ *  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 <memory>
+#include <vector>
+
+#include "webrtc/base/array_view.h"
+#include "webrtc/base/checks.h"
+#include "webrtc/common_audio/channel_buffer.h"
+#include "webrtc/modules/include/module_common_types.h"
+
+namespace webrtc {
+namespace test {
+
+// Abstract class for simulating a microphone with analog gain.
+//
+// The intended mode of operation is the following:
+//
+// auto fake_mic = FakeRecordingDeviceLinear(255);
+//
+// fake_mic.set_mic_level(170);
+// fake_mic.set_mic_level(rtc::Optional<int>());
+// fake_mic.SimulateAnalogGain(buffer);
+//
+// Simulate the microphone level 170.
+//
+// fake_mic.set_mic_level(170);
+// fake_mic.set_mic_level(rtc::Optional<int>(30));
+// fake_mic.SimulateAnalogGain(buffer);
+//
+// Virtually restore the unmodified microphone level knowing that the data in
+// buffer has recorded from a microphone having 30 as level.
+// Then, calling SimulateAnalogGain() will first "undo" the gain applied by the
+// real microphone.
+class FakeRecordingDevice {
+ public:
+  enum class DeviceKind { IDENTITY, LINEAR };
+
+  explicit FakeRecordingDevice(int initial_mic_level);
+  virtual ~FakeRecordingDevice() = 0;
+
+  // FakeRecordingDevice factory.
+  static std::unique_ptr<FakeRecordingDevice> GetFakeRecDevice(
+      DeviceKind kind, int initial_mic_level);
+
+  // Setter and getter for the mic level to simulate.
+  void set_mic_level(int level);
+  int mic_level() const;
+
+  // Setter and getter for the mic level to undo.
+  void set_undo_mic_level(rtc::Optional<int> level);
+  rtc::Optional<int> undo_mic_level() const;
+
+  // Simulates the analog gain on an std::vector<rtc::ArrayView<float>> buffer.
+  // If |real_device_level| is a valid level, the unmodified mic signal is
+  // virtually restored. To skip the latter step set |real_device_level| to
+  // an empty value.
+  void SimulateAnalogGain(std::vector<rtc::ArrayView<float>> buffer);

[AleBzk, 2017/05/23 13:56:41]

@Per: thanks for your comments on keeping the API as essential as possible; I
removed the methods that was only used in the unit tests and adapted the UTs
accordingly.

+
+  // Simulates the analog gain on a ChannelBuffer<float> buffer.
+  // For further details, see the comment above.
+  void SimulateAnalogGain(ChannelBuffer<float>* buffer);
+
+  // Simulates the analog gain on an AudioFrame buffer (fixed point).
+  // For further details, see the comment above.
+  void SimulateAnalogGain(AudioFrame* buffer);
+
+ protected:
+  // Abstract methods required by SimulateAnalogGain().
+  virtual void ModifySampleInt16(int16_t* sample) = 0;
+  virtual void ModifySampleFloat(float* sample) = 0;
+
+  int16_t ClipSampleInt16(int16_t sample);
+  float ClipSampleFloat(float sample);
+
+ private:
+  // Mic level to simulate.
+  int mic_level_;
+
+  // Optional undo mic level.
+  rtc::Optional<int> undo_mic_level_;
+};
+
+}  // namespace test
+}  // namespace webrtc
+
+#endif  // WEBRTC_MODULES_AUDIO_PROCESSING_TEST_FAKE_RECORDING_DEVICE_H_