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使用AVAudioEngine进行电平测量

iosobjective-cmacosavfoundationavaudioengine
23

我刚刚观看了WWDC视频（Session 502 AVAudioEngine 实践），对使用这项技术构建应用程序感到非常兴奋。

到目前为止，我还没有找到如何监控麦克风输入或混音输出级别的方法。

有人可以帮忙吗？ 明确一点，我指的是监控当前输入信号（并在UI中显示它），而不是通道/轨道的输入/输出音量设置。

我知道你可以使用AVAudioRecorder来实现此操作，但这不是AVAudioEngine所需的AVAudioNode。

6个回答
25

尝试在主混合器上安装水龙头，然后通过设置帧长度使其更快，接着读取样本并计算平均值，就像这样:

在顶部导入框架

#import <Accelerate/Accelerate.h>

添加属性

@property float averagePowerForChannel0;
@property float averagePowerForChannel1;

然后是下面的相同>>

self.mainMixer = [self.engine mainMixerNode];
[self.mainMixer installTapOnBus:0 bufferSize:1024 format:[self.mainMixer outputFormatForBus:0] block:^(AVAudioPCMBuffer * _Nonnull buffer, AVAudioTime * _Nonnull when) {
    [buffer setFrameLength:1024];
    UInt32 inNumberFrames = buffer.frameLength;

    if(buffer.format.channelCount>0)
    {
        Float32* samples = (Float32*)buffer.floatChannelData[0];
        Float32 avgValue = 0;

        vDSP_meamgv((Float32*)samples, 1, &avgValue, inNumberFrames);
        self.averagePowerForChannel0 = (LEVEL_LOWPASS_TRIG*((avgValue==0)?-100:20.0*log10f(avgValue))) + ((1-LEVEL_LOWPASS_TRIG)*self.averagePowerForChannel0) ;
        self.averagePowerForChannel1 = self.averagePowerForChannel0;
    }

    if(buffer.format.channelCount>1)
    {
        Float32* samples = (Float32*)buffer.floatChannelData[1];
        Float32 avgValue = 0;

        vDSP_meamgv((Float32*)samples, 1, &avgValue, inNumberFrames);
        self.averagePowerForChannel1 = (LEVEL_LOWPASS_TRIG*((avgValue==0)?-100:20.0*log10f(avgValue))) + ((1-LEVEL_LOWPASS_TRIG)*self.averagePowerForChannel1) ;
    }
}];

然后，获取您想要的目标值

NSLog(@"===test===%.2f", self.averagePowerForChannel1);

为了获得最大峰值，请使用vDSP_maxmgv而不是vDSP_meamgv。

LEVEL_LOWPASS_TRIG是一个简单的过滤器，取值介于0.0到1.0之间。如果您将其设置为0.0，则会过滤掉所有值并且不会得到任何数据。如果您将其设置为1.0，您将获得太多的噪音。基本上，您设置的值越高，您将获得更多的数据变化。对于大多数应用程序，似乎在0.10到0.30之间的值比较好。

7
LEVEL_LOWPASS_TRIG的值（或范围）是多少？ - apocolipse
6
使用 vDSP_meanv，需导入 "Accelerate" 库，使用苹果的高性能数学框架。 - Josh
1
你能否在Github上发布一个完整的可工作示例呢？ - real 19
嗨，Josh，Swift 转换对你来说是否有效？如果是，请分享一下。 - SaRaVaNaN DM
很棒的代码。@FarhadMalekpour，你能否添加更多注释来解释代码在做什么以及为什么这样做吗？ - Mr Rogers
显示剩余4条评论
17

'Farhad Malekpour'的回答的等效Swift 3代码：

在顶部导入框架

import Accelerate

全局声明

private var audioEngine: AVAudioEngine?
    private var averagePowerForChannel0: Float = 0
    private var averagePowerForChannel1: Float = 0
let LEVEL_LOWPASS_TRIG:Float32 = 0.30

在需要的地方使用以下代码

let inputNode = audioEngine!.inputNode//since i need microphone audio level i have used `inputNode` otherwise you have to use `mainMixerNode`
let recordingFormat: AVAudioFormat = inputNode!.outputFormat(forBus: 0)
 inputNode!.installTap(onBus: 0, bufferSize: 1024, format: recordingFormat) {[weak self] (buffer:AVAudioPCMBuffer, when:AVAudioTime) in
                guard let strongSelf = self else {
                    return
                }
                strongSelf.audioMetering(buffer: buffer)
}

计算

private func audioMetering(buffer:AVAudioPCMBuffer) {
            buffer.frameLength = 1024
            let inNumberFrames:UInt = UInt(buffer.frameLength)
            if buffer.format.channelCount > 0 {
                let samples = (buffer.floatChannelData![0])
                var avgValue:Float32 = 0
                vDSP_meamgv(samples,1 , &avgValue, inNumberFrames)
                var v:Float = -100
                if avgValue != 0 {
                    v = 20.0 * log10f(avgValue)
                }
                self.averagePowerForChannel0 = (self.LEVEL_LOWPASS_TRIG*v) + ((1-self.LEVEL_LOWPASS_TRIG)*self.averagePowerForChannel0)
                self.averagePowerForChannel1 = self.averagePowerForChannel0
            }

            if buffer.format.channelCount > 1 {
                let samples = buffer.floatChannelData![1]
                var avgValue:Float32 = 0
                vDSP_meamgv(samples, 1, &avgValue, inNumberFrames)
                var v:Float = -100
                if avgValue != 0 {
                    v = 20.0 * log10f(avgValue)
                }
                self.averagePowerForChannel1 = (self.LEVEL_LOWPASS_TRIG*v) + ((1-self.LEVEL_LOWPASS_TRIG)*self.averagePowerForChannel1)
            }
    }
你有这段代码的工作示例吗？展示整个周期...如何实例化AudioEngine等。 - omarojo
2
新手问题 - 如果节点设置在通道0上，为什么会有2个通道？ - Starwave
2

Swift 5+

我从这个项目中得到了帮助。

  1. download above project & copy 'Microphone.swift' class in your project.

  2. copy paste these fowling codes in your project:

    import AVFoundation

private var mic = MicrophoneMonitor(numberOfSamples: 1)
private var timer:Timer!

override func viewDidLoad() {
    super.viewDidLoad()
    timer = Timer.scheduledTimer(timeInterval: 0.1, target: self, selector: #selector(startMonitoring), userInfo: nil, repeats: true)
    timer.fire()
}

@objc func startMonitoring() {
  print("sound level:", normalizeSoundLevel(level: mic.soundSamples.first!))
}

private func normalizeSoundLevel(level: Float) -> CGFloat {
    let level = max(0.2, CGFloat(level) + 50) / 2 // between 0.1 and 25
    return CGFloat(level * (300 / 25)) // scaled to max at 300 (our height of our bar)
}

3. 打开一瓶啤酒，庆祝一下吧！

1
这是在不断地将音频录制到文件中吗？看起来并不是很高效。 - omarojo
这是我找到的唯一方法！ - Ahmadreza
1
我发现了另一种解决方案，有点奇怪，但比tap功能更好。混音器没有AudioUnit，但是如果将其转换为AVAudioIONode，就可以获得AudioUnit并使用iOS的计量设备。以下是操作方法：
启用或禁用计量：
- (void)setMeteringEnabled:(BOOL)enabled;
{
    UInt32 on = (enabled)?1:0;
    AVAudioIONode *node = (AVAudioIONode*)self.engine.mainMixerNode;
    OSStatus err = AudioUnitSetProperty(node.audioUnit, kAudioUnitProperty_MeteringMode, kAudioUnitScope_Output, 0, &on, sizeof(on));
}

更新计量表：
- (void)updateMeters;
{
    AVAudioIONode *node = (AVAudioIONode*)self.engine.mainMixerNode;

    AudioUnitParameterValue level;
    AudioUnitGetParameter(node.audioUnit, kMultiChannelMixerParam_PostAveragePower, kAudioUnitScope_Output, 0, &level);

    self.averagePowerForChannel1 = self.averagePowerForChannel0 = level;
    if(self.numberOfChannels>1)
    {
        err = AudioUnitGetParameter(node.audioUnit, kMultiChannelMixerParam_PostAveragePower+1, kAudioUnitScope_Output, 0, &level);
    }
}
1
#define LEVEL_LOWPASS_TRIG .3

#import "AudioRecorder.h"





@implementation AudioRecord


-(id)init {
     self = [super init];
     self.recordEngine = [[AVAudioEngine alloc] init];

     return self;
}


 /**  ----------------------  Snippet Stackoverflow.com not including Audio Level Meter    ---------------------     **/


-(BOOL)recordToFile:(NSString*)filePath {

     NSURL *fileURL = [NSURL fileURLWithPath:filePath];

     const Float64 sampleRate = 44100;

     AudioStreamBasicDescription aacDesc = { 0 };
     aacDesc.mSampleRate = sampleRate;
     aacDesc.mFormatID = kAudioFormatMPEG4AAC; 
     aacDesc.mFramesPerPacket = 1024;
     aacDesc.mChannelsPerFrame = 2;

     ExtAudioFileRef eaf;

     OSStatus err = ExtAudioFileCreateWithURL((__bridge CFURLRef)fileURL, kAudioFileAAC_ADTSType, &aacDesc, NULL, kAudioFileFlags_EraseFile, &eaf);
     assert(noErr == err);

     AVAudioInputNode *input = self.recordEngine.inputNode;
     const AVAudioNodeBus bus = 0;

     AVAudioFormat *micFormat = [input inputFormatForBus:bus];

     err = ExtAudioFileSetProperty(eaf, kExtAudioFileProperty_ClientDataFormat, sizeof(AudioStreamBasicDescription), micFormat.streamDescription);
     assert(noErr == err);

     [input installTapOnBus:bus bufferSize:1024 format:micFormat block:^(AVAudioPCMBuffer *buffer, AVAudioTime *when) {
       const AudioBufferList *abl = buffer.audioBufferList;
       OSStatus err = ExtAudioFileWrite(eaf, buffer.frameLength, abl);
       assert(noErr == err);


       /**  ----------------------  Snippet from stackoverflow.com in different context  ---------------------     **/


       UInt32 inNumberFrames = buffer.frameLength;
       if(buffer.format.channelCount>0) {
         Float32* samples = (Float32*)buffer.floatChannelData[0]; 
         Float32 avgValue = 0;
         vDSP_maxv((Float32*)samples, 1.0, &avgValue, inNumberFrames);
         self.averagePowerForChannel0 = (LEVEL_LOWPASS_TRIG*((avgValue==0)?
                                  -100:20.0*log10f(avgValue))) + ((1- LEVEL_LOWPASS_TRIG)*self.averagePowerForChannel0) ;
         self.averagePowerForChannel1 = self.averagePowerForChannel0;
       }

       dispatch_async(dispatch_get_main_queue(), ^{

         self.levelIndicator.floatValue=self.averagePowerForChannel0;

       });     


       /**  ---------------------- End of Snippet from stackoverflow.com in different context  ---------------------     **/

     }];

     BOOL startSuccess;
     NSError *error;

     startSuccess = [self.recordEngine startAndReturnError:&error]; 
     return startSuccess;
}



@end
针对@omarojo，这是一个使用两个其他答案组合的可行代码。.h文件即将到来。 - Paul-J
-1
#import <Foundation/Foundation.h>
#import <AVFoundation/AVFoundation.h>
#import <AudioToolbox/ExtendedAudioFile.h>
#import <CoreAudio/CoreAudio.h>
#import <Accelerate/Accelerate.h>
#import <AppKit/AppKit.h>

@interface AudioRecord : NSObject {

}

@property (nonatomic) AVAudioEngine *recordEngine;


@property float averagePowerForChannel0;
@property float averagePowerForChannel1;
@property float numberOfChannels;
@property NSLevelIndicator * levelIndicator;


-(BOOL)recordToFile:(NSString*)filePath;

@end
1
使用方法很简单，只需调用以下代码：newAudioRecord = [AudioRecord new]; newAudioRecord.levelIndicator=self.audioLevelIndicator; --- 实验性的（不是很好）[newAudioRecord recordToFile:fullFilePath_Name]; [newAudioRecord.recordEngine stop]; [newAudioRecord.recordEngine reset];newAudioRecord.recordEngine pause];要恢复录音，请使用以下代码：[newAudioRecord.recordEngine startAndReturnError:NULL]; - Paul-J
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