webrtc-AudioprcessingModule 3A算法demo

以2019年代码为例；


//
//  Created by yizhimao on 2020.
// 

#import "ViewController.h"

#import "ViewController.h"
#include "audio_processing.h"
#include"module_common_types.h"


@interface ViewController ()

@property webrtc::AudioProcessing* apm;

@end

///////////////////////////////////
 bool is_echo_cancel = false;
  FILE *echo_in = NULL;
 int  delay_ms = -1;
//////////////////////////////////////
@implementation ViewController

- (void)viewDidLoad {
    [super viewDidLoad];
   
    [ self setparam:2 NAME:"AEC"];
    [self createAudio];
    // Do any additional setup after loading the view.
}
-(int)setparam:(int) level NAME:(char *)arg{

    // Usage example, omitting error checking:
    _apm = webrtc::AudioProcessing::Create();
    _apm->high_pass_filter()->Enable(true);
    if(arg == "ANR")
    {
        switch (level) {
                  case 0:
                   _apm->noise_suppression()->set_level(webrtc::NoiseSuppression::kLow);
                      break;
                  case 1:
                   _apm->noise_suppression()->set_level(webrtc::NoiseSuppression::kModerate);
                      break;
                  case 2:
                   _apm->noise_suppression()->set_level(webrtc::NoiseSuppression::kHigh);
                      break;
                  case 3:
                   _apm->noise_suppression()->set_level(webrtc::NoiseSuppression::kVeryHigh);
                      break;
                  default:
                   _apm->noise_suppression()->set_level(webrtc::NoiseSuppression::kVeryHigh);
                  }
           
            _apm->noise_suppression()->Enable(true);//噪声抑制组件，client必须启用
           //----------------------------------------------------------------------------------
    }else if (arg == "AGC") {
       
        _apm->gain_control()->set_analog_level_limits(0, 100);
        _apm->gain_control()->set_mode(webrtc::GainControl::kAdaptiveDigital);//模拟信号字自适应调节
        //apm->gain_control()->set_mode(webrtc::GainControl::kAdaptiveAnalog);
        //Default Gain control mode is kAdaptiveAnalog, so if mode is not set,
        //that means kAdaptiveAnalog is being used.
        
        _apm->gain_control()->Enable(true);
    } else if (arg == "AEC") {
        webrtc::EchoCancellation *echo_cancell = _apm->echo_cancellation();
      
        echo_cancell->enable_drift_compensation(false);//d是否启用补偿机制
        switch (level) {//回声抑制增强等级
            case 0:
                echo_cancell->set_suppression_level(webrtc::EchoCancellation::kLowSuppression);
                break;
            case 1:
                echo_cancell->set_suppression_level(webrtc::EchoCancellation::kModerateSuppression);
                break;
            case 2:
                echo_cancell->set_suppression_level(webrtc::EchoCancellation::kHighSuppression);
        }
        is_echo_cancel  = true;
        delay_ms = 0;
        ////////////////////////////
        NSArray *dicts = NSSearchPathForDirectoriesInDomains(NSDocumentDirectory, NSUserDomainMask, YES);
               NSString *path = [dicts objectAtIndex:0];
               NSLog(@"the path is %@", path);
               
            echo_in  = fopen([[NSString stringWithFormat:@"%@/Captureccref.pcm", path] UTF8String], "rb");//参考信号
        if(echo_in == NULL)
        {
            return -1;
        }
        ////////////////////////////////////////
        echo_cancell->Enable(true);
        _apm->gain_control()->Enable(false);
        _apm->noise_suppression()->Enable(false);
    
    }else {
        delete _apm;
       
    }
  //  _apm->level_estimator()->Enable(true);//启用重试次数估计组件

  //  _apm->echo_cancellation()->Enable(true);//启用回声消除组件

   // _apm->echo_cancellation()->enable_metrics(true);//允许计算回波指标

  //  _apm->echo_cancellation()->enable_drift_compensation(true);
    //启用时钟补偿模块（声音捕捉设备的时钟频率和播放设备的时钟频率可能不一样）

   // _apm->gain_control()->Enable(true);//启用增益控制组件，client必须启用哦！

   // _apm->high_pass_filter()->Enable(true);//高通过滤器组件，过滤DC偏移和低频噪音，client必须启用


   // _apm->voice_detection()->Enable(true);//启用语音检测组件，检测是否有说话声

   // _apm->voice_detection()->set_likelihood( webrtc::VoiceDetection::kModerateLikelihood);//设置语音检测的阀值，阀值越大，语音越不容易被忽略，同样一些噪音可能被当成语音。

    _apm->Initialize();//保留所有用户设置的情况下重新初始化apm的内部状态，用于开始处理一个新的音频流。第一个流创建之后不一定需要调用此方法。
    
    
    return 0;
}
bool ReadFrame(FILE* file, webrtc::AudioFrame* frame) {
    // The files always contain stereo audio.
    size_t frame_size = frame->samples_per_channel_;
    size_t read_count = fread(frame->data_,
                              sizeof(int16_t),//以short的形式读取，16位精度
                              frame_size,
                              file);
    if (read_count != frame_size) {
        // Check that the file really ended.
        return false;  // This is expected.
    }
    return true;
}
//
bool WriteFrame(FILE* file, webrtc::AudioFrame* frame) {
    // The files always contain stereo audio.
    size_t frame_size = frame->samples_per_channel_;
    size_t read_count = fwrite (frame->data_,
                                sizeof(int16_t),
                                frame_size,
                                file);
    if (read_count != frame_size) {
        return false;  // This is expected.
    }
    return true;
}
-(void)createAudio{
      
    webrtc::AudioFrame *frame = new webrtc::AudioFrame();
      float frame_step = 10;  // ms
      frame->sample_rate_hz_ = 8000;
      frame->samples_per_channel_ = frame->sample_rate_hz_ * frame_step / 1000.0;//10毫秒采集多少采样点（后边读数据会按照精度转成short来读）

      frame->num_channels_ = 1;
      webrtc::AudioFrame *echo_frame = NULL;
      if (is_echo_cancel) {
          echo_frame = new webrtc::AudioFrame();
          echo_frame->sample_rate_hz_ = 8000;
          echo_frame->samples_per_channel_ = echo_frame->sample_rate_hz_ * frame_step / 1000.0;

          echo_frame->num_channels_ = 1;
      }
       //-------
    static  FILE *wav_in = NULL;
    static  FILE *wav_out = NULL;
    if (wav_in == NULL ||wav_out == NULL )
    {
        NSArray *dicts = NSSearchPathForDirectoriesInDomains(NSDocumentDirectory, NSUserDomainMask, YES);
        NSString *path = [dicts objectAtIndex:0];
        NSLog(@"the path is %@", path);
        
       wav_in  = fopen([[NSString stringWithFormat:@"%@/Capturecc.pcm", path] UTF8String], "rb");
        wav_out  = fopen([[NSString stringWithFormat:@"%@/waveoutAEC.pcm", path] UTF8String], "wb");
    }
    //----------
   if (wav_in == NULL ||wav_out == NULL )
   {
       return ;
   }
    
    
   int num_frame = 0;
   while (ReadFrame(wav_in, frame)) {
       num_frame += 1;
       if (is_echo_cancel) {
           if (ReadFrame(echo_in, echo_frame)) {
               _apm->ProcessReverseStream(echo_frame);//分析处理远端信号（参考信号，渲染信号）
               _apm->set_stream_delay_ms(10);//apm->set_stream_delay_ms(0);设置延时值
           }
       }
//       if(_apm->gain_control()->is_enabled()) {//           //apm->gain_control()->set_stream_analog_level(level);
//           //apm->gain_control()->set_target_level_dbfs(level);
//       }
       _apm->ProcessStream(frame);//处理近端(采集)数据;如果需要启用的功能 任何带有set_stream_tag标记的函数必须在处理当前帧之前调用;包括各个环节的处理。（如增益调节、回声消除、噪声抑制、语音检测、高通过率等，没有解码！是针对pcm数据做处理的）
       WriteFrame(wav_out, frame);
   }

 

       fclose(wav_in);
       fclose(wav_out);
       if(is_echo_cancel && echo_in)
           fclose(echo_in);

       if(frame)
       {
           delete frame;
           frame = NULL;
       }
       if (is_echo_cancel)
           delete echo_frame;

    if(_apm)
       {
           delete _apm;
           _apm = NULL;
       }
}

@end

 

 我们扣一些文件来测测它的3A算法