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..b26a552d9012692f92e08b006ae9a3bf28c7c23c
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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/base/optional.h"
+#include "webrtc/common_audio/channel_buffer.h"
+#include "webrtc/modules/include/module_common_types.h"
+
+namespace webrtc {
+namespace test {
+
+class FakeRecordingDeviceWorker;
+
+// Class for simulating a microphone with analog gain.
+//
+// The intended modes of operation are the following:
+//
+// FakeRecordingDevice fake_mic(255, 1);
+//
+// fake_mic.set_mic_level(170);
+// fake_mic.set_undo_mic_level(rtc::Optional<int>());
+// fake_mic.SimulateAnalogGain(buffer);
+//
+// When the mic level to undo is known:
+//
+// fake_mic.set_mic_level(170);
+// fake_mic.set_undo_mic_level(rtc::Optional<int>(30));
+// fake_mic.SimulateAnalogGain(buffer);
+//
+// The second option virtually restores the unmodified microphone level. Calling
+// SimulateAnalogGain() will first "undo" the gain applied by the real
+// microphone (e.g., 30).
+class FakeRecordingDevice final {
+ public:
+  FakeRecordingDevice(int initial_mic_level, int device_kind);

[AleBzk, 2017/07/26 13:42:31]

As you requested, this simplifies the client code - no need to cast int to enum.

+  ~FakeRecordingDevice();
+
+  int mic_level() const { return mic_level_; }
+
+  void set_mic_level(int level) { mic_level_ = level; }
+  void set_undo_mic_level(rtc::Optional<int> level) { undo_mic_level_ = level; }
+
+  // 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);
+
+  // 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);
+
+ private:
+  // Mic level to simulate.
+  int mic_level_;
+
+  // Fake recording device worker.
+  std::unique_ptr<FakeRecordingDeviceWorker> worker_;
+
+  // Optional undo mic level.
+  rtc::Optional<int> undo_mic_level_;
+};
+
+}  // namespace test
+}  // namespace webrtc
+
+#endif  // WEBRTC_MODULES_AUDIO_PROCESSING_TEST_FAKE_RECORDING_DEVICE_H_