| /* |
| * 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 <vector> |
| |
| #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<int> 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_ |