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Re: Basic question
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Re: Basic question


  • Subject: Re: Basic question
  • From: Steve Bird <email@hidden>
  • Date: Wed, 27 Mar 2002 14:27:24 -0500

On Wednesday, March 27, 2002, at 06:38 AM, Matthew Johnson wrote:

The information I have retrieved is:

input mSampleRate = 44100
input mFormatFlags = 0000000B
input mBytesPerPacket = 8
input mFramesPerPacket = 1
input mChannelsPerFrame = 2
input mBytesPerFrame = 8
input mBitsPerChannel = 32

What I don't understand is what each float value actually is.

--- At the risk of revealing the true depths of my ignorance - I'll jump into this.

--- I would guess (g-u-e-s-s = I haven't yet played with this) that the floats have a theoretical range of +/- 0.50000,
possibly +/- 1.0000. That means they are a fraction of full scale. What full scale is (in terms of volts) is a function of the hardware, and might or might not be tightly controlled from one machine to the next, or from one model to another.

--- One way to find this out is to whistle loudly very close into the microphone, and examine the data. It should show you the limits fairly plainly, since the signal will be clipped. Most of the data will be either +full scale, or -full scale - only a few points in between.


Can I
determine from each value ( which I am assuming is a sample ) the Amplitude
in Db's and the frequency in Hz.
--- You can calculate a dB level from one sample - but it's a meaningless number. Normal use of dB as a loudness meter requires you to calculate the RMS value of the signal over time (multiple samples) and use that.

--- You cannot calculate the frequency of a signal from one sample. There is no such thing as "the" frequency of a signal, unless that signal is a 100% pure sine wave. Most signals are composite, i.e. made up of multiple frequencies. A common technique for looking at this is the Fast Fourier Transform (FFT). That's a procedure for transforming a block of time samples into a block of amplitude-at-different-frequency values. Find a text on basic signal processing for this.


I.e if I record some audio I seem to get floats within the range of -0.4 to
0.4 (I don't know what the units are) As far as I can deduce each value must
represent a Amplitude and a Frequency.

I could be completely wrong.
--- I think you are completely wrong in that.


I am a total newbie to this.
--- I was once, also. My experience comes from 20+ years of handing signals, but I started where you are.

----------------------------------------------------------------
Steve Bird
Culverson Software - Elegant software that is a pleasure to use.
www.Culverson.com (toll free) 1-877-676-8175
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  • Follow-Ups:
    • Re: Basic question
      • From: Jeff Moore <email@hidden>
References: 
 >Re: Basic question (From: Matthew Johnson <email@hidden>)

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