WEBRTC系列文章：

WEBRTC系列之基于IOS平台编译WEBRTC（一）

1、AGC 初识

    自动增益控制电路的作用是：当输入信号电压变化很大时，保持接收机输出电压恒定或基本不变。具体地说，当输入信号很弱时，接收机的增益大，自动增益控制电路不起作用；当输入信号很强时，自动增益控制电路进行控制，使接收机的增益减小。这样，当接收信号强度变化时，接收机的输出端的电压或功率基本不变或保持恒定。因此对AGC电路的要求是：在输入信号较小时，AGC电路不起作用，只有当输入信号增大到一定程度后，AGC电路才起控制作用，使增益随输入信号的增大而减少。
    为实现上述要求，必须有一个能随外来信号强弱而变化的控制电压或电流信号，利用这个信号对放大器的增益自动进行控制。由上述分析可知，调幅中频信号经幅度检波后，在它的输出中除音频信号外，还含有直流分量。直流分量大小与中频载波的振幅成正比，也即与外来高频信号成正比。因此，可将检波器输出的直流分量作为AGC控制信号。

    AGC电路工作原理：可以分为增益受控放大电路和控制电压形成电路。增益受控放大电路位于正向放大通路，其增益随控制电压U0而改变。控制电压形成电路的基本部件是AGC整流器和低通平滑滤波器，有时也包含门电路和直流放大器等部件。

2、WEBRTC 的 AGC算法

    AGC是自动增益补偿功能（Automatic Gain Control），AGC可以自动调麦克风的收音量，使与会者收到一定的音量水平，不会因发言者与麦克风的距离改变时，声音有忽大忽小声的缺点。webbrtc中的结构如下：

说明：gain_control.h是包装的头文件，在apm里头gain_control_impl调用。主要包括了接口定义函数和参数配置。

3、主要配置

• include/gain_control.h 里面定义了agc的三种模式，kAdaptiveAnalog、kAdaptiveDigital和kFixedDigital。其中，kAdaptiveAnalog带有模拟音量调节的功能。kAdaptiveDigital是可变增益agc，但是不调节系统音量。kFixedDigital是固定增益的agc。

enum Mode {
    // Adaptive mode intended for use if an analog volume control is available
    // on the capture device. It will require the user to provide coupling
    // between the OS mixer controls and AGC through the |stream_analog_level()|
    // functions.
    //
    // It consists of an analog gain prescription for the audio device and a
    // digital compression stage.
    kAdaptiveAnalog,

    // Adaptive mode intended for situations in which an analog volume control
    // is unavailable. It operates in a similar fashion to the adaptive analog
    // mode, but with scaling instead applied in the digital domain. As with
    // the analog mode, it additionally uses a digital compression stage.
    kAdaptiveDigital,

    // Fixed mode which enables only the digital compression stage also used by
    // the two adaptive modes.
    //
    // It is distinguished from the adaptive modes by considering only a
    // short time-window of the input signal. It applies a fixed gain through
    // most of the input level range, and compresses (gradually reduces gain
    // with increasing level) the input signal at higher levels. This mode is
    // preferred on embedded devices where the capture signal level is
    // predictable, so that a known gain can be applied.
    kFixedDigital
  };

• analog_agc.h 定义了配置targetLevelDbfs和compressionGaindB用于调节agc的动态范围

typedef struct {
  // Configurable parameters/variables
  uint32_t fs;                // Sampling frequency
  int16_t compressionGaindB;  // Fixed gain level in dB
  int16_t targetLevelDbfs;    // Target level in -dBfs of envelope (default -3)
  int16_t agcMode;            // Hard coded mode (adaptAna/adaptDig/fixedDig)
  uint8_t limiterEnable;      // Enabling limiter (on/off (default off))
  WebRtcAgcConfig defaultConfig;
  WebRtcAgcConfig usedConfig;

  // General variables
  int16_t initFlag;
  int16_t lastError;

  // Target level parameters
  // Based on the above: analogTargetLevel = round((32767*10^(-22/20))^2*16/2^7)
  int32_t analogTargetLevel;    // = RXX_BUFFER_LEN * 846805;       -22 dBfs
  int32_t startUpperLimit;      // = RXX_BUFFER_LEN * 1066064;      -21 dBfs
  int32_t startLowerLimit;      // = RXX_BUFFER_LEN * 672641;       -23 dBfs
  int32_t upperPrimaryLimit;    // = RXX_BUFFER_LEN * 1342095;      -20 dBfs
  int32_t lowerPrimaryLimit;    // = RXX_BUFFER_LEN * 534298;       -24 dBfs
  int32_t upperSecondaryLimit;  // = RXX_BUFFER_LEN * 2677832;      -17 dBfs
  int32_t lowerSecondaryLimit;  // = RXX_BUFFER_LEN * 267783;       -27 dBfs
  uint16_t targetIdx;           // Table index for corresponding target level
#ifdef MIC_LEVEL_FEEDBACK
  uint16_t targetIdxOffset;  // Table index offset for level compensation
#endif
  int16_t analogTarget;  // Digital reference level in ENV scale

  // Analog AGC specific variables
  int32_t filterState[8];  // For downsampling wb to nb
  int32_t upperLimit;      // Upper limit for mic energy
  int32_t lowerLimit;      // Lower limit for mic energy
  int32_t Rxx160w32;       // Average energy for one frame
  int32_t Rxx16_LPw32;     // Low pass filtered subframe energies
  int32_t Rxx160_LPw32;    // Low pass filtered frame energies
  int32_t Rxx16_LPw32Max;  // Keeps track of largest energy subframe
  int32_t Rxx16_vectorw32[RXX_BUFFER_LEN];  // Array with subframe energies
  int32_t Rxx16w32_array[2][5];  // Energy values of microphone signal
  int32_t env[2][10];            // Envelope values of subframes

  int16_t Rxx16pos;          // Current position in the Rxx16_vectorw32
  int16_t envSum;            // Filtered scaled envelope in subframes
  int16_t vadThreshold;      // Threshold for VAD decision
  int16_t inActive;          // Inactive time in milliseconds
  int16_t msTooLow;          // Milliseconds of speech at a too low level
  int16_t msTooHigh;         // Milliseconds of speech at a too high level
  int16_t changeToSlowMode;  // Change to slow mode after some time at target
  int16_t firstCall;         // First call to the process-function
  int16_t msZero;            // Milliseconds of zero input
  int16_t msecSpeechOuterChange;  // Min ms of speech between volume changes
  int16_t msecSpeechInnerChange;  // Min ms of speech between volume changes
  int16_t activeSpeech;           // Milliseconds of active speech
  int16_t muteGuardMs;            // Counter to prevent mute action
  int16_t inQueue;                // 10 ms batch indicator

  // Microphone level variables
  int32_t micRef;         // Remember ref. mic level for virtual mic
  uint16_t gainTableIdx;  // Current position in virtual gain table
  int32_t micGainIdx;     // Gain index of mic level to increase slowly
  int32_t micVol;         // Remember volume between frames
  int32_t maxLevel;       // Max possible vol level, incl dig gain
  int32_t maxAnalog;      // Maximum possible analog volume level
  int32_t maxInit;        // Initial value of "max"
  int32_t minLevel;       // Minimum possible volume level
  int32_t minOutput;      // Minimum output volume level
  int32_t zeroCtrlMax;    // Remember max gain => don't amp low input
  int32_t lastInMicLevel;

  int16_t scale;  // Scale factor for internal volume levels
#ifdef MIC_LEVEL_FEEDBACK
  int16_t numBlocksMicLvlSat;
  uint8_t micLvlSat;
#endif
  // Structs for VAD and digital_agc
  AgcVad vadMic;
  DigitalAgc digitalAgc;

#ifdef WEBRTC_AGC_DEBUG_DUMP
  FILE* fpt;
  FILE* agcLog;
  int32_t fcount;
#endif

  int16_t lowLevelSignal;
} LegacyAgc;

相关源代码如下：

int GainControlImpl::AnalyzeCaptureAudio(AudioBuffer* audio) {
  if (!enabled_) {
    return AudioProcessing::kNoError;
  }

  RTC_DCHECK(num_proc_channels_);
  RTC_DCHECK_GE(AudioBuffer::kMaxSplitFrameLength,
                audio->num_frames_per_band());
  RTC_DCHECK_EQ(audio->num_channels(), *num_proc_channels_);
  RTC_DCHECK_LE(*num_proc_channels_, gain_controllers_.size());

  int16_t split_band_data[AudioBuffer::kMaxNumBands]
                         [AudioBuffer::kMaxSplitFrameLength];
  int16_t* split_bands[AudioBuffer::kMaxNumBands] = {
      split_band_data[0], split_band_data[1], split_band_data[2]};

  if (mode_ == kAdaptiveAnalog) {
    int capture_channel = 0;
    for (auto& gain_controller : gain_controllers_) {
      gain_controller->set_capture_level(analog_capture_level_);

      audio->ExportSplitChannelData(capture_channel, split_bands);

      int err =
          WebRtcAgc_AddMic(gain_controller->state(), split_bands,
                           audio->num_bands(), audio->num_frames_per_band());

      audio->ImportSplitChannelData(capture_channel, split_bands);

      if (err != AudioProcessing::kNoError) {
        return AudioProcessing::kUnspecifiedError;
      }
      ++capture_channel;
    }
  } else if (mode_ == kAdaptiveDigital) {
    int capture_channel = 0;
    for (auto& gain_controller : gain_controllers_) {
      int32_t capture_level_out = 0;

      audio->ExportSplitChannelData(capture_channel, split_bands);

      int err =
          WebRtcAgc_VirtualMic(gain_controller->state(), split_bands,
                               audio->num_bands(), audio->num_frames_per_band(),
                               analog_capture_level_, &capture_level_out);

      audio->ImportSplitChannelData(capture_channel, split_bands);

      gain_controller->set_capture_level(capture_level_out);

      if (err != AudioProcessing::kNoError) {
        return AudioProcessing::kUnspecifiedError;
      }
      ++capture_channel;
    }
  }

  return AudioProcessing::kNoError;
}

4、主要接口

    analog_agc.h包括模拟的agc结构体声明，而gain_control.h中的接口函数在analog_agc.c中实现。

WebRtcAgc_AddFarend  计算远端信号的语音活度VAD
WebRtcAgc_AddMic 计算麦克风输入的语音活度，对于非常小的信号会乘增益系数
WebRtcAgc_VirtualMic	用虚拟的麦克风音量来调节幅度
WebRtcAgc_Process	vad核心处理
WebRtcAgc_set_config	设置VAD参数

在analog_agc.c还包括以下函数：

WebRtcAgc_UpdateAgcThresholds	
WebRtcAgc_SaturationCtrl	
WebRtcAgc_ZeroCtrl	
WebRtcAgc_SpeakerInactiveCtrl	
WebRtcAgc_ExpCurve	
WebRtcAgc_ProcessAnalog	

digital_agc.h包括数字的agc结构体声明，Vad结构声明，而gain_control.h中的接口函数在analog_agc.c中实现。

WebRtcAgc_ProcessDigital	
WebRtcAgc_AddFarendToDigital	
WebRtcAgc_InitVad	
WebRtcAgc_ProcessVad	
WebRtcAgc_CalculateGainTable	

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最后

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