| /* |
| * Copyright (c) 2019 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 MODULES_AUDIO_PROCESSING_GAIN_CONTROL_CONFIG_PROXY_H_ |
| #define MODULES_AUDIO_PROCESSING_GAIN_CONTROL_CONFIG_PROXY_H_ |
| |
| #include "modules/audio_processing/include/audio_processing.h" |
| #include "modules/audio_processing/include/gain_control.h" |
| #include "rtc_base/critical_section.h" |
| #include "rtc_base/thread_annotations.h" |
| |
| namespace webrtc { |
| |
| // This class forwards all gain control configuration to the audio processing |
| // module, for compatibility with AudioProcessing::Config. |
| class GainControlConfigProxy : public GainControl { |
| public: |
| GainControlConfigProxy(rtc::CriticalSection* crit_capture, |
| AudioProcessing* apm, |
| GainControl* agc); |
| GainControlConfigProxy(const GainControlConfigProxy&) = delete; |
| GainControlConfigProxy& operator=(const GainControlConfigProxy&) = delete; |
| |
| ~GainControlConfigProxy() override; |
| |
| private: |
| // GainControl API during processing. |
| int set_stream_analog_level(int level) override; |
| int stream_analog_level() const override; |
| |
| // GainControl config setters. |
| int Enable(bool enable) override; |
| int set_mode(Mode mode) override; |
| int set_target_level_dbfs(int level) override; |
| int set_compression_gain_db(int gain) override; |
| int enable_limiter(bool enable) override; |
| int set_analog_level_limits(int minimum, int maximum) override; |
| |
| // GainControl config getters. |
| bool is_enabled() const override; |
| bool is_limiter_enabled() const override; |
| int compression_gain_db() const override; |
| int target_level_dbfs() const override; |
| int analog_level_minimum() const override; |
| int analog_level_maximum() const override; |
| bool stream_is_saturated() const override; |
| Mode mode() const override; |
| |
| rtc::CriticalSection* crit_capture_ = nullptr; |
| AudioProcessing* apm_ = nullptr; |
| GainControl* agc_ RTC_GUARDED_BY(crit_capture_) = nullptr; |
| }; |
| |
| } // namespace webrtc |
| #endif // MODULES_AUDIO_PROCESSING_GAIN_CONTROL_CONFIG_PROXY_H_ |