我做正确的事情将分贝从-120 - 0转换为0 - 120

Des*_*ond 8 iphone xcode audio-recording decibel ios

我想测量周围的音量,不太确定我做的是否正确.

我想创建一个范围为0(安静)到120(非常嘈杂)的VU表.

我获得了峰值和平均功率,但在正常安静的环境中非常高.请给我一些指针.

- (void)viewDidLoad
{
    [super viewDidLoad];
    // Do any additional setup after loading the view, typically from a nib.


    //creating an audio CAF file in the temporary directory, this isn’t ideal but it’s the only way to get this class functioning (the temporary directory is erased once the app quits). Here we also specifying a sample rate of 44.1kHz (which is capable of representing 22 kHz of sound frequencies according to the Nyquist theorem), and 1 channel (we do not need stereo to measure noise).

    NSDictionary* recorderSettings = [NSDictionary dictionaryWithObjectsAndKeys:
                                      [NSNumber numberWithInt:kAudioFormatLinearPCM],AVFormatIDKey,
                                      [NSNumber numberWithInt:44100],AVSampleRateKey,
                                      [NSNumber numberWithInt:1],AVNumberOfChannelsKey,
                                      [NSNumber numberWithInt:16],AVLinearPCMBitDepthKey,
                                      [NSNumber numberWithBool:NO],AVLinearPCMIsBigEndianKey,
                                      [NSNumber numberWithBool:NO],AVLinearPCMIsFloatKey,
                                      nil];
    NSError* error;

    NSURL *url = [NSURL fileURLWithPath:@"/dev/null"];
    recorder = [[AVAudioRecorder alloc] initWithURL:url settings:recorderSettings error:&error];

    //enable measuring
    //tell the recorder to start recording:
    [recorder record];

    if (recorder) {
        [recorder prepareToRecord];
        recorder.meteringEnabled = YES;
        [recorder record];
        levelTimer = [NSTimer scheduledTimerWithTimeInterval: 0.01 target: self selector: @selector(levelTimerCallback:) userInfo: nil repeats: YES];

    } else
    {
        NSLog(@"%@",[error description]);
    }        
}

- (void)levelTimerCallback:(NSTimer *)timer {
    [recorder updateMeters];

    const double ALPHA = 0.05;
    double peakPowerForChannel = pow(10, (0.05 * [recorder averagePowerForChannel:0]));
    lowPassResults = ALPHA * peakPowerForChannel + (1.0 - ALPHA) * lowPassResults;     

    NSLog(@"Average input: %f Peak input: %f Low pass results: %f", [recorder averagePowerForChannel:0], [recorder peakPowerForChannel:0], lowPassResults);

    float tavgPow =[recorder averagePowerForChannel:0] + 120.0;
    float tpPow = [recorder peakPowerForChannel:0] + 120.0;

    float avgPow = tavgPow;//(float)abs([recorder averagePowerForChannel:0]);
    float pPow = tpPow;//(float)abs([recorder peakPowerForChannel:0]);

    NSString *tempAvg = [NSString stringWithFormat:@"%0.2f",avgPow];
        NSString *temppeak = [NSString stringWithFormat:@"%0.2f",pPow];
    [avg setText:tempAvg];
        [peak setText:temppeak];
    NSLog(@"Average input: %f Peak input: %f Low pass results: %f", avgPow,pPow , lowPassResults);
}   
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mea*_*ers 25

Apple在其SpeakHere示例中使用查找表,该表将dB转换为电平表上显示的线性值.这是为了节省设备功率(我猜).

我也需要这个,但是并不认为每1/10秒(我的刷新率)进行一次浮点计算会花费很多设备功率.所以,我没有建立一个表格,而是将代码模拟成:

float       level;                // The linear 0.0 .. 1.0 value we need.
const float minDecibels = -80.0f; // Or use -60dB, which I measured in a silent room.
float       decibels    = [audioRecorder averagePowerForChannel:0];

if (decibels < minDecibels)
{
    level = 0.0f;
}
else if (decibels >= 0.0f)
{
    level = 1.0f;
}
else
{
    float   root            = 2.0f;
    float   minAmp          = powf(10.0f, 0.05f * minDecibels);
    float   inverseAmpRange = 1.0f / (1.0f - minAmp);
    float   amp             = powf(10.0f, 0.05f * decibels);
    float   adjAmp          = (amp - minAmp) * inverseAmpRange;

    level = powf(adjAmp, 1.0f / root);
}
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我正在使用AVAudioRecorder,因此您可以看到dB averagePowerForChannel:,但您可以在那里填写自己的dB值.

Apple的例子使用了double计算,我不明白,因为音频计量float精度绰绰有余,而且设备功耗更低.

毋庸置疑,您现在可以level通过简单的方法将此计算范围缩放到0 ... 120范围level * 120.0f.

上面的代码可以加快当我们固定root2.0f,通过更换powf(adjAmp, 1.0f / root)具有sqrtf(adjAmp); 但这是一个小问题,一个非常好的编译器可能能够为我们这样做.我几乎肯定inverseAmpRange会在编译时计算一次.


Mic*_*nen 8

当你想使用1.0作为参考(对于0db)时,将线性幅度转换为分贝的公式是

20 * log10(amp);
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因此,我不确定查看代码的意图,但您可能想要

float db = 20 * log10([recorder averagePowerForChannel:0]);
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这将从幅度为零的-infinity到幅度为1的0db.如果你真的需要它在0到120之间,你可以添加120并在零处使用最大函数.

所以,经过以上这一行:

db += 120;
db = db < 0 ? 0 : db;
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您使用的公式似乎是将DB转换为放大器的公式,我认为这与您想要的相反.

编辑:我重读,似乎你可能已经有了分贝值.

如果是这种情况,只需不要转换为幅度并添加120.

所以改变

double peakPowerForChannel = pow(10, (0.05 * [recorder averagePowerForChannel:0]));
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double peakPowerForChannel = [recorder averagePowerForChannel:0];
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你应该没事的去吧.