/* * Copyright (c) 2012 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_IMPL_H_ #define WEBRTC_MODULES_AUDIO_PROCESSING_GAIN_CONTROL_IMPL_H_ #include #include "webrtc/modules/audio_processing/include/audio_processing.h" #include "webrtc/modules/audio_processing/processing_component.h" namespace webrtc { class AudioBuffer; class CriticalSectionWrapper; class GainControlImpl : public GainControl, public ProcessingComponent { public: GainControlImpl(const AudioProcessing* apm, CriticalSectionWrapper* crit); virtual ~GainControlImpl(); int ProcessRenderAudio(AudioBuffer* audio); int AnalyzeCaptureAudio(AudioBuffer* audio); int ProcessCaptureAudio(AudioBuffer* audio); // ProcessingComponent implementation. int Initialize() override; // GainControl implementation. bool is_enabled() const override; int stream_analog_level() override; bool is_limiter_enabled() const override; Mode mode() const override; private: // GainControl implementation. int Enable(bool enable) override; int set_stream_analog_level(int level) override; int set_mode(Mode mode) 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; int set_analog_level_limits(int minimum, int maximum) override; int analog_level_minimum() const override; int analog_level_maximum() const override; bool stream_is_saturated() const override; // ProcessingComponent implementation. void* CreateHandle() const override; int InitializeHandle(void* handle) const override; int ConfigureHandle(void* handle) const override; void DestroyHandle(void* handle) const override; int num_handles_required() const override; int GetHandleError(void* handle) const override; const AudioProcessing* apm_; CriticalSectionWrapper* crit_; Mode mode_; int minimum_capture_level_; int maximum_capture_level_; bool limiter_enabled_; int target_level_dbfs_; int compression_gain_db_; std::vector capture_levels_; int analog_capture_level_; bool was_analog_level_set_; bool stream_is_saturated_; }; } // namespace webrtc #endif // WEBRTC_MODULES_AUDIO_PROCESSING_GAIN_CONTROL_IMPL_H_