WEBRTC系列之AGC 算法原理初始（三）

Webrtc 系列文章：

WEBRTC系列之基于IOS平台编译WEBRTC（一）

WEBRTC系列之 AGC 算法原理初识（二）

1、 基本概念

1.1、 IOS平台自带AGC功能

    可以通过以下方法查看ios平台是否支持AGC和设置AGC（代码摘自webrtc）：

//1. 创建Component，设置开启硬件音频处理
// Create an audio component description to identify the Voice Processing
  // I/O audio unit.
  AudioComponentDescription vpio_unit_description;
  vpio_unit_description.componentType = kAudioUnitType_Output;
  vpio_unit_description.componentSubType = kAudioUnitSubType_VoiceProcessingIO;//设置这个 sub type 后，会启用系统的 AEC, AGC, NS 等
  vpio_unit_description.componentManufacturer = kAudioUnitManufacturer_Apple;
  vpio_unit_description.componentFlags = 0;
  vpio_unit_description.componentFlagsMask = 0;

  // Obtain an audio unit instance given the description.
  AudioComponent found_vpio_unit_ref =
      AudioComponentFindNext(nullptr, &vpio_unit_description);

  // Create a Voice Processing IO audio unit.
  OSStatus result = noErr;
  result = AudioComponentInstanceNew(found_vpio_unit_ref, &vpio_unit_);
  if (result != noErr) {
    vpio_unit_ = nullptr;
    RTCLogError(@"AudioComponentInstanceNew failed. Error=%ld.", (long)result);
    return false;
  }


// 2. 查看是否支持AGC，webRTC中封装在GetAGCState方法中
UInt32 size = sizeof(*enabled);
  OSStatus result = AudioUnitGetProperty(vpio_unit_,
                                         kAUVoiceIOProperty_VoiceProcessingEnableAGC,
                                         kAudioUnitScope_Global,
                                         kInputBus,
                                         enabled,
                                         &size);
  RTCLog(@"VPIO unit AGC: %u", static_cast<unsigned int>(*enabled));

  // 设置开启AGC
    UInt32 enable_agc = 1;
    result =
        AudioUnitSetProperty(vpio_unit_,
                             kAUVoiceIOProperty_VoiceProcessingEnableAGC,
                             kAudioUnitScope_Global, kInputBus, &enable_agc,
                             sizeof(enable_agc));
    if (result != noErr) {
      RTCLogError(@"Failed to enable the built-in AGC. "
                   "Error=%ld.",
                  (long)result);
      RTC_HISTOGRAM_COUNTS_SPARSE_100000(
          "WebRTC.Audio.SetAGCStateErrorCode", (-1) * result);
    }

//3. 初始化AudioUnit
AudioUnitInitialize(vpio_unit_);

1.2、关键回调接口分析

    在直播录制中比较关键的一步就是Render Callback Function（webrtc中对应设置的方法为OnDeliverRecordedData）。AudioUnit每次都是处理一段音频数据，每次处理完成一段数据的时候，这个回调函数就会被调用一次。 代码摘自webrtc：

/ Disable AU buffer allocation for the recorder, we allocate our own.
  // TODO(henrika): not sure that it actually saves resource to make this call.
  UInt32 flag = 0;
  result = AudioUnitSetProperty(
      vpio_unit_, kAudioUnitProperty_ShouldAllocateBuffer,
      kAudioUnitScope_Output, kInputBus, &flag, sizeof(flag));
  if (result != noErr) {
    DisposeAudioUnit();
    RTCLogError(@"Failed to disable buffer allocation on the input bus. "
                 "Error=%ld.",
                (long)result);
    return false;
  }

  // Specify the callback to be called by the I/O thread to us when input audio
  // is available. The recorded samples can then be obtained by calling the
  // AudioUnitRender() method.
  AURenderCallbackStruct input_callback;
  input_callback.inputProc = OnDeliverRecordedData;
  input_callback.inputProcRefCon = this;
  result = AudioUnitSetProperty(vpio_unit_,
                                kAudioOutputUnitProperty_SetInputCallback,
                                kAudioUnitScope_Global, kInputBus,