audioproc_f with simulated mic analog gain

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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index 0000000000000000000000000000000000000000..1db4d97962cbf494420040c688b7f55a85e029d5
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+++ 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 <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 {

[peah-webrtc, 2017/05/23 22:13:20]

 I really think we should put as little implementation as possible into this
class now, i.e., make it as small and simple as possible. The reason for that is
that we don't really know how to simulate this. 

I'm open to having one or two simple default implementations just to show how it
is supposed to work but I don't think it makes sense to do advanced
implementations in a specific way. For instance, it is fairly clear to me that
the current approach won't be sufficient to simulate a microphone with a noise
suppressor (which we have on Macbooks), beamformers, or a microphone with any
type of memory in its clipping behavior. Until we can rule out that we don't
need such an implementation of the simulation I don't think we should have a
code design that enforces that, unless that code is very simple.

Why not do this as simple as possible, and not more than that? It will be much
easier to discuss the implementation once we really know what is needed to
simulate the device.

[AleBzk, 2017/06/22 10:16:00]

Done.

+ 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.
+  // 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(ChannelBuffer<float>* buffer);
+
+  // Simulates the analog gain.
+  // 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(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_