|  | /* | 
|  | *  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 MODULES_AUDIO_PROCESSING_GAIN_CONTROL_IMPL_H_ | 
|  | #define MODULES_AUDIO_PROCESSING_GAIN_CONTROL_IMPL_H_ | 
|  |  | 
|  | #include <stddef.h> | 
|  | #include <stdint.h> | 
|  |  | 
|  | #include <memory> | 
|  | #include <optional> | 
|  | #include <vector> | 
|  |  | 
|  | #include "api/array_view.h" | 
|  | #include "modules/audio_processing/agc/gain_control.h" | 
|  |  | 
|  | namespace webrtc { | 
|  |  | 
|  | class ApmDataDumper; | 
|  | class AudioBuffer; | 
|  |  | 
|  | class GainControlImpl : public GainControl { | 
|  | public: | 
|  | GainControlImpl(); | 
|  | GainControlImpl(const GainControlImpl&) = delete; | 
|  | GainControlImpl& operator=(const GainControlImpl&) = delete; | 
|  |  | 
|  | ~GainControlImpl() override; | 
|  |  | 
|  | void ProcessRenderAudio(ArrayView<const int16_t> packed_render_audio); | 
|  | int AnalyzeCaptureAudio(const AudioBuffer& audio); | 
|  | int ProcessCaptureAudio(AudioBuffer* audio, bool stream_has_echo); | 
|  |  | 
|  | void Initialize(size_t num_proc_channels, int sample_rate_hz); | 
|  |  | 
|  | static void PackRenderAudioBuffer(const AudioBuffer& audio, | 
|  | std::vector<int16_t>* packed_buffer); | 
|  |  | 
|  | // GainControl implementation. | 
|  | int stream_analog_level() const override; | 
|  | bool is_limiter_enabled() const override { return limiter_enabled_; } | 
|  | Mode mode() const override { return mode_; } | 
|  | int set_mode(Mode mode) override; | 
|  | int compression_gain_db() const override { return compression_gain_db_; } | 
|  | int set_analog_level_limits(int minimum, int maximum) override; | 
|  | int set_compression_gain_db(int gain) override; | 
|  | int set_target_level_dbfs(int level) override; | 
|  | int enable_limiter(bool enable) override; | 
|  | int set_stream_analog_level(int level) override; | 
|  |  | 
|  | private: | 
|  | struct MonoAgcState; | 
|  |  | 
|  | // GainControl implementation. | 
|  | int target_level_dbfs() const override { return target_level_dbfs_; } | 
|  | int analog_level_minimum() const override { return minimum_capture_level_; } | 
|  | int analog_level_maximum() const override { return maximum_capture_level_; } | 
|  | bool stream_is_saturated() const override { return stream_is_saturated_; } | 
|  |  | 
|  | int Configure(); | 
|  |  | 
|  | std::unique_ptr<ApmDataDumper> data_dumper_; | 
|  |  | 
|  | Mode mode_; | 
|  | int minimum_capture_level_; | 
|  | int maximum_capture_level_; | 
|  | bool limiter_enabled_; | 
|  | int target_level_dbfs_; | 
|  | int compression_gain_db_; | 
|  | int analog_capture_level_ = 0; | 
|  | bool was_analog_level_set_; | 
|  | bool stream_is_saturated_; | 
|  |  | 
|  | std::vector<std::unique_ptr<MonoAgcState>> mono_agcs_; | 
|  | std::vector<int> capture_levels_; | 
|  |  | 
|  | std::optional<size_t> num_proc_channels_; | 
|  | std::optional<int> sample_rate_hz_; | 
|  |  | 
|  | static int instance_counter_; | 
|  | }; | 
|  | }  // namespace webrtc | 
|  |  | 
|  | #endif  // MODULES_AUDIO_PROCESSING_GAIN_CONTROL_IMPL_H_ |