/* * Copyright (c) 2013 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_AGC_AGC_MANAGER_DIRECT_H_ #define MODULES_AUDIO_PROCESSING_AGC_AGC_MANAGER_DIRECT_H_ #include #include "modules/audio_processing/agc/agc.h" #include "modules/audio_processing/logging/apm_data_dumper.h" #include "rtc_base/constructormagic.h" #include "rtc_base/gtest_prod_util.h" namespace webrtc { class AudioFrame; class DebugFile; class GainControl; // Callbacks that need to be injected into AgcManagerDirect to read and control // the volume values. This is done to remove the VoiceEngine dependency in // AgcManagerDirect. // TODO(aluebs): Remove VolumeCallbacks. class VolumeCallbacks { public: virtual ~VolumeCallbacks() {} virtual void SetMicVolume(int volume) = 0; virtual int GetMicVolume() = 0; }; // Direct interface to use AGC to set volume and compression values. // AudioProcessing uses this interface directly to integrate the callback-less // AGC. // // This class is not thread-safe. class AgcManagerDirect final { public: // AgcManagerDirect will configure GainControl internally. The user is // responsible for processing the audio using it after the call to Process. // The operating range of startup_min_level is [12, 255] and any input value // outside that range will be clamped. AgcManagerDirect(GainControl* gctrl, VolumeCallbacks* volume_callbacks, int startup_min_level, int clipped_level_min, bool use_agc2_level_estimation, bool disable_digital_adaptive); ~AgcManagerDirect(); int Initialize(); void AnalyzePreProcess(int16_t* audio, int num_channels, size_t samples_per_channel); void Process(const int16_t* audio, size_t length, int sample_rate_hz); // Call when the capture stream has been muted/unmuted. This causes the // manager to disregard all incoming audio; chances are good it's background // noise to which we'd like to avoid adapting. void SetCaptureMuted(bool muted); bool capture_muted() { return capture_muted_; } float voice_probability(); private: friend class AgcManagerDirectTest; FRIEND_TEST_ALL_PREFIXES(AgcManagerDirectStandaloneTest, DisableDigitalDisablesDigital); // Dependency injection for testing. Don't delete |agc| as the memory is owned // by the manager. AgcManagerDirect(Agc* agc, GainControl* gctrl, VolumeCallbacks* volume_callbacks, int startup_min_level, int clipped_level_min); // Most general c-tor. AgcManagerDirect(Agc* agc, GainControl* gctrl, VolumeCallbacks* volume_callbacks, int startup_min_level, int clipped_level_min, bool use_agc2_level_estimation, bool disable_digital_adaptive); // Sets a new microphone level, after first checking that it hasn't been // updated by the user, in which case no action is taken. void SetLevel(int new_level); // Set the maximum level the AGC is allowed to apply. Also updates the // maximum compression gain to compensate. The level must be at least // |kClippedLevelMin|. void SetMaxLevel(int level); int CheckVolumeAndReset(); void UpdateGain(); void UpdateCompressor(); std::unique_ptr data_dumper_; static int instance_counter_; std::unique_ptr agc_; GainControl* gctrl_; VolumeCallbacks* volume_callbacks_; int frames_since_clipped_; int level_; int max_level_; int max_compression_gain_; int target_compression_; int compression_; float compression_accumulator_; bool capture_muted_; bool check_volume_on_next_process_; bool startup_; const bool use_agc2_level_estimation_; const bool disable_digital_adaptive_; int startup_min_level_; const int clipped_level_min_; int calls_since_last_gain_log_ = 0; std::unique_ptr file_preproc_; std::unique_ptr file_postproc_; RTC_DISALLOW_COPY_AND_ASSIGN(AgcManagerDirect); }; } // namespace webrtc #endif // MODULES_AUDIO_PROCESSING_AGC_AGC_MANAGER_DIRECT_H_