1 /* 2 * Copyright (c) 2013 The WebRTC project authors. All Rights Reserved. 3 * 4 * Use of this source code is governed by a BSD-style license 5 * that can be found in the LICENSE file in the root of the source 6 * tree. An additional intellectual property rights grant can be found 7 * in the file PATENTS. All contributing project authors may 8 * be found in the AUTHORS file in the root of the source tree. 9 */ 10 11 #ifndef WEBRTC_MODULES_AUDIO_PROCESSING_AGC_AGC_MANAGER_DIRECT_H_ 12 #define WEBRTC_MODULES_AUDIO_PROCESSING_AGC_AGC_MANAGER_DIRECT_H_ 13 14 #include "webrtc/base/scoped_ptr.h" 15 #include "webrtc/modules/audio_processing/agc/agc.h" 16 17 namespace webrtc { 18 19 class AudioFrame; 20 class DebugFile; 21 class GainControl; 22 23 // Callbacks that need to be injected into AgcManagerDirect to read and control 24 // the volume values. This is done to remove the VoiceEngine dependency in 25 // AgcManagerDirect. 26 // TODO(aluebs): Remove VolumeCallbacks. 27 class VolumeCallbacks { 28 public: ~VolumeCallbacks()29 virtual ~VolumeCallbacks() {} 30 virtual void SetMicVolume(int volume) = 0; 31 virtual int GetMicVolume() = 0; 32 }; 33 34 // Direct interface to use AGC to set volume and compression values. 35 // AudioProcessing uses this interface directly to integrate the callback-less 36 // AGC. 37 // 38 // This class is not thread-safe. 39 class AgcManagerDirect final { 40 public: 41 // AgcManagerDirect will configure GainControl internally. The user is 42 // responsible for processing the audio using it after the call to Process. 43 // The operating range of startup_min_level is [12, 255] and any input value 44 // outside that range will be clamped. 45 AgcManagerDirect(GainControl* gctrl, 46 VolumeCallbacks* volume_callbacks, 47 int startup_min_level); 48 // Dependency injection for testing. Don't delete |agc| as the memory is owned 49 // by the manager. 50 AgcManagerDirect(Agc* agc, 51 GainControl* gctrl, 52 VolumeCallbacks* volume_callbacks, 53 int startup_min_level); 54 ~AgcManagerDirect(); 55 56 int Initialize(); 57 void AnalyzePreProcess(int16_t* audio, 58 int num_channels, 59 size_t samples_per_channel); 60 void Process(const int16_t* audio, size_t length, int sample_rate_hz); 61 62 // Call when the capture stream has been muted/unmuted. This causes the 63 // manager to disregard all incoming audio; chances are good it's background 64 // noise to which we'd like to avoid adapting. 65 void SetCaptureMuted(bool muted); capture_muted()66 bool capture_muted() { return capture_muted_; } 67 68 float voice_probability(); 69 70 private: 71 // Sets a new microphone level, after first checking that it hasn't been 72 // updated by the user, in which case no action is taken. 73 void SetLevel(int new_level); 74 75 // Set the maximum level the AGC is allowed to apply. Also updates the 76 // maximum compression gain to compensate. The level must be at least 77 // |kClippedLevelMin|. 78 void SetMaxLevel(int level); 79 80 int CheckVolumeAndReset(); 81 void UpdateGain(); 82 void UpdateCompressor(); 83 84 rtc::scoped_ptr<Agc> agc_; 85 GainControl* gctrl_; 86 VolumeCallbacks* volume_callbacks_; 87 88 int frames_since_clipped_; 89 int level_; 90 int max_level_; 91 int max_compression_gain_; 92 int target_compression_; 93 int compression_; 94 float compression_accumulator_; 95 bool capture_muted_; 96 bool check_volume_on_next_process_; 97 bool startup_; 98 int startup_min_level_; 99 100 rtc::scoped_ptr<DebugFile> file_preproc_; 101 rtc::scoped_ptr<DebugFile> file_postproc_; 102 103 RTC_DISALLOW_COPY_AND_ASSIGN(AgcManagerDirect); 104 }; 105 106 } // namespace webrtc 107 108 #endif // WEBRTC_MODULES_AUDIO_PROCESSING_AGC_AGC_MANAGER_DIRECT_H_ 109