| /* |
| * 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 <memory> |
| |
| #include "modules/audio_processing/agc/agc.h" |
| #include "rtc_base/constructormagic.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); |
| // 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); |
| ~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: |
| // 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<Agc> 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_; |
| int startup_min_level_; |
| const int clipped_level_min_; |
| |
| std::unique_ptr<DebugFile> file_preproc_; |
| std::unique_ptr<DebugFile> file_postproc_; |
| |
| RTC_DISALLOW_COPY_AND_ASSIGN(AgcManagerDirect); |
| }; |
| |
| } // namespace webrtc |
| |
| #endif // MODULES_AUDIO_PROCESSING_AGC_AGC_MANAGER_DIRECT_H_ |