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..c49da4c4bf6621b86f395f1c7e3bd143bb793efb |
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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/common_audio/channel_buffer.h" |
+#include "webrtc/modules/include/module_common_types.h" |
+#include "webrtc/rtc_base/array_view.h" |
+#include "webrtc/rtc_base/checks.h" |
+#include "webrtc/rtc_base/optional.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.SetMicLevel(170); |
+// fake_mic.SetUndoMicLevel(rtc::Optional<int>()); |
+// fake_mic.SimulateAnalogGain(buffer); |
+// |
+// When the mic level to undo is known: |
+// |
+// fake_mic.SetMicLevel(170); |
+// fake_mic.SetUndoMicLevel(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); |
+ ~FakeRecordingDevice(); |
+ |
+ int MicLevel() const; |
+ void SetMicLevel(const int level); |
+ void SetUndoMicLevel(const rtc::Optional<int> 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: |
+ // Fake recording device worker. |
+ std::unique_ptr<FakeRecordingDeviceWorker> worker_; |
+}; |
+ |
+} // namespace test |
+} // namespace webrtc |
+ |
+#endif // WEBRTC_MODULES_AUDIO_PROCESSING_TEST_FAKE_RECORDING_DEVICE_H_ |