| Index: webrtc/modules/audio_processing/gain_control_for_experimental_agc.h
|
| diff --git a/webrtc/modules/audio_processing/gain_control_for_experimental_agc.h b/webrtc/modules/audio_processing/gain_control_for_experimental_agc.h
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| new file mode 100644
|
| index 0000000000000000000000000000000000000000..4fbd05c6850964824efbe9c195c102cfdfd81a52
|
| --- /dev/null
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| +++ b/webrtc/modules/audio_processing/gain_control_for_experimental_agc.h
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| @@ -0,0 +1,70 @@
|
| +/*
|
| + * Copyright (c) 2016 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_GAIN_CONTROL_FOR_EXPERIMENTAL_AGC_H_
|
| +#define WEBRTC_MODULES_AUDIO_PROCESSING_GAIN_CONTROL_FOR_EXPERIMENTAL_AGC_H_
|
| +
|
| +#include "webrtc/base/constructormagic.h"
|
| +#include "webrtc/base/criticalsection.h"
|
| +#include "webrtc/base/thread_checker.h"
|
| +#include "webrtc/modules/audio_processing/agc/agc_manager_direct.h"
|
| +#include "webrtc/modules/audio_processing/include/audio_processing.h"
|
| +
|
| +namespace webrtc {
|
| +
|
| +// This class has two main purposes:
|
| +//
|
| +// 1) It is returned instead of the real GainControl after the new AGC has been
|
| +// enabled in order to prevent an outside user from overriding compression
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| +// settings. It doesn't do anything in its implementation, except for
|
| +// delegating the const methods and Enable calls to the real GainControl, so
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| +// AGC can still be disabled.
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| +//
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| +// 2) It is injected into AgcManagerDirect and implements volume callbacks for
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| +// getting and setting the volume level. It just caches this value to be used
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| +// in VoiceEngine later.
|
| +class GainControlForExperimentalAgc : public GainControl,
|
| + public VolumeCallbacks {
|
| + public:
|
| + explicit GainControlForExperimentalAgc(GainControl* gain_control,
|
| + rtc::CriticalSection* crit_capture);
|
| +
|
| + // GainControl implementation.
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| + int Enable(bool enable) override;
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| + bool is_enabled() const override;
|
| + int set_stream_analog_level(int level) override;
|
| + int stream_analog_level() override;
|
| + int set_mode(Mode mode) override;
|
| + Mode mode() const override;
|
| + int set_target_level_dbfs(int level) override;
|
| + int target_level_dbfs() const override;
|
| + int set_compression_gain_db(int gain) override;
|
| + int compression_gain_db() const override;
|
| + int enable_limiter(bool enable) override;
|
| + bool is_limiter_enabled() const override;
|
| + int set_analog_level_limits(int minimum, int maximum) override;
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| + int analog_level_minimum() const override;
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| + int analog_level_maximum() const override;
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| + bool stream_is_saturated() const override;
|
| +
|
| + // VolumeCallbacks implementation.
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| + void SetMicVolume(int volume) override;
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| + int GetMicVolume() override;
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| +
|
| + private:
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| + GainControl* real_gain_control_;
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| + int volume_;
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| + rtc::CriticalSection* crit_capture_;
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| + RTC_DISALLOW_COPY_AND_ASSIGN(GainControlForExperimentalAgc);
|
| +};
|
| +
|
| +} // namespace webrtc
|
| +
|
| +#endif // WEBRTC_MODULES_AUDIO_PROCESSING_GAIN_CONTROL_FOR_EXPERIMENTAL_AGC_H_
|
|
|