我刚刚在AVAudioEngine
上观看了WWDC视频(Session 502 AVAudioEngine
in Practice),我非常兴奋能够基于这项技术制作一个应用程序。
我还没能弄清楚如何对麦克风输入或混音器输出进行电平监控。
有人能帮忙吗?需要说明的是,我说的是监控当前输入信号(并在UI中显示),而不是通道/音轨的输入/输出音量设置。
我知道你可以用AVAudioRecorder
这样做,但这不是AVAudioEngine
需要的AVAudioNode
。
尝试在主混音器上安装一个水龙头,然后通过设置帧长使其更快,然后读取样本并获得平均值,如下所示:
导入框架
#import <Accelerate/Accelerate.h>
添加属性
@property float averagePowerForChannel0;
@property float averagePowerForChannel1;
则下面相同>>
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,得到你想要的目标值
NSLog(@"===test===%.2f", self.averagePowerForChannel1);
获取峰值,使用vDSP_maxmgv而不是vDSP_meamgv。
LEVEL_LOWPASS_TRIG是一个值在0.0到1.0之间的简单过滤器,如果你设置0.0,你将过滤所有值而不获得任何数据。如果你把它设置为1.0,你会得到太多的噪音。基本上,该值越高,数据的变化就越大。对于大多数应用程序来说,0.10到0.30之间的值似乎比较合适。
'Farhad Malekpour'的等效Swift 3代码的回答
导入框架
import Accelerate
宣布在全球范围内
private var audioEngine: AVAudioEngine?
private var averagePowerForChannel0: Float = 0
private var averagePowerForChannel1: Float = 0
let LEVEL_LOWPASS_TRIG:Float32 = 0.30
在需要的地方使用下面的代码
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)
}
}
Swift 5+
我从这个项目中得到了帮助。
-
下载以上项目&
-
复制粘贴这些代码到你的项目中:
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。开瓶啤酒庆祝!
我发现了另一个有点奇怪的解决方案,但效果很好,比tap要好得多。混频器没有AudioUnit,但如果你将它转换为AVAudioIONode,你可以获得AudioUnit并使用iOS的计量功能。方法如下:
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));
}
更新仪表:
- (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);
}
}
#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
#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