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Multimedia PeakMeter Control

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4 Sep 2008CPOL6 min read 316.3K   22.3K   222   116
Multimedia PeakMeter control - .NET version
PeakMeter Application

soundsmfc.jpg

soundsmfc.jpg

Introduction

A while ago, I released the .NET version of the Peak Meter control but one thing I quickly discovered is that many developers would not know how to use it in their own application. This is the reason for this update. I will present the various components that are involved in order to show audio meter in real-time.

Description

Real-time Peak Meter processing is composed of three blocks.

Mic processing
  • The source: Source of audio data, this can be a microphone, a digital sound file (CD audio) or even a video if you consider grabbing the audio part of it. Whatever your source the principle is the same, you will have to convert the data to a format that is suitable for Signal Processing.
  • DSP Processing: What I'm calling DSP processing block here is the component that is responsible to convert the audio data to peak amplitude format.
  • Peak Meter rendering: This is basically the rendering control (just like the Peak Meter control of this article). The rendering engine doesn't have to be all fancy like this control. A progress control can be used to show Peak audio. In fact, if you look around Windows control panel, this is exactly what is being used.

Audio Source

The microphone is the most basic audio source that you can think of.
Basically the previous figure illustrates how your computer receives digital samples of the sound that is produced from the microphone. The way the whole process works is that you specify the format that you wish to receive the data, mostly PCM data (Pulse Code Modulation) and the audio capture device will collect the samples based on sampling frequency. PCM data and WAVE chunks are common terms to describe the data that is obtained from the ADC (Analog to Digital Converter).
For example to digitize audio samples from a microphone at 22050 Hz single channel audio (mono) and 16 bits resolution; and let's say you want to receive samples every 100 ms, you will have to provide a buffer of 4410 bytes (22050*(16/8)*0.1).

I mentioned previously about file source. A file source is simply an audio data storage that gives you quick access to your samples. Audio formats like: MP3, WMA and Ogg Vorbis; all they do is compressing the data while trying to maintain the audio quality.

DSP Processing

DSP processing is a very interesting subject to learn and work with. This block receives digital samples from the source. It approximates the original waveform and finds its peak magnitudes.
Since I would not be able to go in details about how FFT (Fast Fourier Transform) works in this article, I recommend the interested reader to visit some of the links in the reference section to increase his/her knowledge about this process.

FFT plays an important role in signal processing and is probably one of the most written subjects in software engineering. When dealing with digital system, the ADC (Analog to Digital Converter) gives us a set of digital audio samples (discrete signal). The theory behind it tells us that when we perform a DFT (Discrete Fourier Transform) on a discrete signal, we find its composant frequencies including their phase and amplitude. I would like to direct the advanced reader that I'm not talking only about pure sine wave where only the fundamental is visible but performing a DFT (or FFT) on audio signal generally produces some amplitudes on nearby bin frequencies.

Now all we have left to do is find the range of frequencies for our analysis. According to the sampling theorem we can approximate the maximum audio frequency in our signal. The Nyquist theorem states that the baseband Fs > 2B, which means the sampling frequency (digital) must be at least twice any frequency in the range of B frequencies (analog) in order to reconstruct the original signal and prevent aliasing. Aliasing is the effect that causes a different signal to be indistinguishable (or aliases of one another) when sampled. Thus, if we sample at 44.1KHz (audio CD quality), we could capture the entire range of sound (roughly 20KHz).

Note that the sampling frequency needs to be at least twice our max frequency, using 44.1KHz to digitize a 20KHz sine waves improves our spectrum analysis by getting a few more samples than necessary.

Ideal FFT
Ideal FFT

Peak Meter Rendering

Peak Meter is just a "piece of cake" since the DSP block takes all the burden of what has to be done. What we do next is that we select a group of frequencies and display the amplitude of their closest bin frequencies. Peak Meter control can be as simple as using a progress control. But I guess we like to be fancy from time to time! The PeakMeter control presented here does just that.

It is best to choose frequencies in the range below because they are typically found in normal conversation and music.

Audio Frequency map

Using PeakMeter in .NET

You can use the .NET version of this control in WinForms applications. Just add it to your toolbox and drag and drop it to your WinForm canvas. Here's a summary of the properties for this control:

PeakMeterControl Properties

PropertyDescription
MeterStyleShow Meter bands in Horizontal (PMS_Horizontal) or Vertical (PMS_Vertical)
ShowGridShow meter background grid
ColoredGridShow meter background grid using color schemes
GridColorBackground grid color (when ColoredGrid property is false)
ColorNormalLow range color
ColorNormalBackLow range color Background (ColoredGrid=1)
ColorMediumMedium range color
ColorMediumBackMedium range color Background (ColoredGrid=1)
ColorHighHigh range color
ColorHighBackHigh range color Background (ColoredGrid=1)
BandsCountNumber of bands
LEDCountNumber of LED per bands (when 1, band is smooth)
FalloffSpeedFalloff effect speed (1=fast, 10=normal, 100=slow)
FalloffEffectEnable falloff effect, call Start() to run
FalloffColorFalloff line color

PeakMeterControl Methods

NameDescription
bool Start(int delay)Start animation (delay in ms). Can be called from non-UI thread.
bool Stop()Stop animation. Can be called from non-UI thread.
void SetMeterBands(int bands, int led)Set meter bands properties and number of LED in one call.
void SetRange(int min, int med, int max)Change meter control default range.
bool SetData(int[] values, int offset, int size)Set peak meter data. Can be called from non-UI thread.

Using the PeakMeter

Using this control is very straightforward, add it to your toolbox and drop it to your form. You can use the Start method to animate the falloff effect. The demo fills the data with random numbers.

C#
private void FillMeterData()
{
    int[] meters1 = new int[NumLEDS];
    Random rand = new Random();
    for (int i = 0; i < meters1.Length; i++)
    {
        meters1[i] = rand.Next(0, 100);
    }
    // fill meter data
    this.peakMeterCtrl1.SetData(meters1, 0, meters1.Length);
}

SoundStudio Demo

SoundStudio Application is a simple sound player application capable of playing various audio files (.wav, .mp3, and .wma). It uses the WindowsMedia .NET library to parse the audio.
SoundStudioCS (C# version) can be built using the unmanaged DLL (FFTLib.dll). Simply add USE_FFTLIB to your project build settings (Build->General tab).

Some of the features of WindowsMedia .NET (Release) are:

  • Microsoft WAVE device support (waveInOpen, waveOutOpen)
  • DirectShow Filters, including sample grabber filter
  • Microsoft Multimedia Streaming Interfaces

References

History

  • 09/04/2008: Updated all samples (+WindowsMedia release)
  • 08/13/2008: SoundStudio Release (include WindowsMedia pre-release)
  • 05/24/2008: First revision for .NET

License

This article, along with any associated source code and files, is licensed under The Code Project Open License (CPOL)


Written By
Software Developer (Senior)
United States United States
Ernest is a multi-discipline software engineer.
Skilled at software design and development for all Windows platforms.
-
MCSD (C#, .NET)
Interests: User Interface, GDI/GDI+, Scripting, Android, iOS, Windows Mobile.
Programming Skills: C/C++, C#, Java (Android), VB and ASP.NET.

I hope you will enjoy my contributions.

Comments and Discussions

 
GeneralRe: Excellent Pin
Te-Deum27-Aug-08 22:08
Te-Deum27-Aug-08 22:08 
GeneralRe: Excellent Pin
Ernest Laurentin28-Aug-08 3:27
Ernest Laurentin28-Aug-08 3:27 
GeneralRe: Excellent Pin
Te-Deum28-Aug-08 7:35
Te-Deum28-Aug-08 7:35 
Questioni upload kj dsp source code, what difference process PCM data between your work and it? Pin
jackyxinli26-Aug-08 23:08
jackyxinli26-Aug-08 23:08 
Questioncan you introduce me an open source sound library like bass? Pin
jackyxinli26-Aug-08 16:35
jackyxinli26-Aug-08 16:35 
AnswerRe: can you introduce me an open source sound library like bass? Pin
Ernest Laurentin26-Aug-08 17:14
Ernest Laurentin26-Aug-08 17:14 
GeneralRe: can you introduce me an open source sound library like bass? Pin
jackyxinli26-Aug-08 17:26
jackyxinli26-Aug-08 17:26 
Questioni got two questiones about function CSoundStudioDlg::GetAudioData Pin
jackyxinli25-Aug-08 21:41
jackyxinli25-Aug-08 21:41 
i got two questiones about function CSoundStudioDlg::GetAudioData, assume WaveFormat.wBitsPerSample=16, WaveFormat.nChannels=2.
1. why did you get right channel first, not left?
2. i got a program which process PCM data in java, and PCM data is a byte array, it's way is:
int c = 0;
for ( int a = 0; a < sampleSize; ) {
if (c >= audioDataBuffer.length) {
offset++;
c -= audioDataBuffer.length;
}

left[a] = (float) ((audioDataBuffer[c + 1] << 8) + audioDataBuffer[c]) / 32767F;
right[a] = (float) ((audioDataBuffer[c + 3] << 8) + audioDataBuffer[c + 2]) / 32767F;
a++;
c += 4;
}

and i convert it to C code below:
const byte* pfData = reinterpret_cast<const byte*>(pbData);
for (size_t i = 0, j=0; i < Samples; ++i, j+=4)
{
//m_RealIn_RT[i] = static_cast<float>( pfData[j] );
//m_RealIn_LT[i] = static_cast<float>( pfData[j+1] );

m_RealIn_RT[i] = (float)( pfData[j+1]<<8 + pfData[j] ) / 32767;
m_RealIn_LT[i] = (float)( pfData[j+3]<<8 + pfData[j+2] ) / 32767;
}

and run it, the spectrum(every band) higher than before, why?
AnswerRe: i got two questiones about function CSoundStudioDlg::GetAudioData Pin
Ernest Laurentin26-Aug-08 3:30
Ernest Laurentin26-Aug-08 3:30 
GeneralA question Pin
jackyxinli14-Aug-08 14:28
jackyxinli14-Aug-08 14:28 
AnswerRe: A question Pin
Ernest Laurentin14-Aug-08 15:03
Ernest Laurentin14-Aug-08 15:03 
GeneralRe: A question Pin
jackyxinli14-Aug-08 15:45
jackyxinli14-Aug-08 15:45 
QuestionThis control in Silverlight? Pin
jones_m1-Aug-08 1:20
jones_m1-Aug-08 1:20 
AnswerRe: This control in Silverlight? Pin
Ernest Laurentin1-Aug-08 13:40
Ernest Laurentin1-Aug-08 13:40 
GeneralRe: This control in Silverlight? Pin
jones_m1-Aug-08 20:47
jones_m1-Aug-08 20:47 
GeneralRe: This control in Silverlight? Pin
Ernest Laurentin13-Aug-08 7:00
Ernest Laurentin13-Aug-08 7:00 
AnswerRe: This control in Silverlight? Pin
_elli_13-Aug-08 20:24
_elli_13-Aug-08 20:24 
GeneralRe: This control in Silverlight? Pin
neoandrew5-Oct-09 11:08
neoandrew5-Oct-09 11:08 
GeneralRe: This control in Silverlight? Pin
_elli_13-Oct-09 0:13
_elli_13-Oct-09 0:13 
QuestionWhat peak does this control record. Pin
Gautam Sharma26-May-08 1:31
Gautam Sharma26-May-08 1:31 
AnswerRe: What peak does this control record. Pin
Axel Rietschin26-May-08 1:51
professionalAxel Rietschin26-May-08 1:51 
GeneralRe: What peak does this control record. Pin
lebunne26-May-08 22:58
lebunne26-May-08 22:58 
GeneralRe: What peak does this control record. Pin
Ernest Laurentin27-May-08 4:46
Ernest Laurentin27-May-08 4:46 
GeneralRe: What peak does this control record. Pin
lebunne27-May-08 21:22
lebunne27-May-08 21:22 
General[Message Removed] Pin
Mojtaba Vali25-May-08 17:52
Mojtaba Vali25-May-08 17:52 

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