Sound

1. What is the limitation of digital audio ?
Audio sampling is based on the Nyquist-Shannon sampling theorem, which states that
for accurate signal reconstruction the sampling frequency should be at least twice the
bandwidth of the source signal.
Therefore, if a source signal has a bandwidth of 0 - 1KHz, then the sampling rate needs
to be at least 2KHz .
Therefore, in keeping with the Nyquist-Shannon theorem the sampling rate needed to
capture
all of the audio in this range needs to be approximately 40KHz, although it should be
pointed out that
many recordings may not have content that conatains frequencies at the higehr or
lower end of this range). Due to the practicalities of building these electrical systems the
most commonly used baseline sample rate is 44.1 KHz systems. 48KHz, 96KHz, and
192KHz are becoming more commonly used as the CD is starting to be replaced by
media capable of supporting high sample rates.
2. What is the characteristics of audio signal?

Audio signals may be characterized by parameters such as Bandwidth , Decibel , PCM ,
sample and hold
1- Bandwidth
The range of available frequencies for a specified signal or system. For
example, a source file containing frequencies from 100Hz to 15KHz has a 14.9KHz
bandwidth between the two frequencies.
2- Decibel
The decibel is a measurement of the relationship between sound intensity and
perceived loudness by the listener.
3- PCM
Pulse Code Modulation (PCM). In terms of digital audio files a linear PCM scheme stores
binary integer values for each sample.
4-(Sample-and-hold)
A technique used in electronics when interfacing with real world inputs. A sample is
taken with most commonly the voltage reading being stored within a capacitor. A switch
disconnects the capacitor from the input, and thus the 12/14 sampled value is held. At
the next period of the sampling frequency the switch is opened, allowing for the input
value to be read again then closed,repeating the process.
3. How to convert between sound and electricity ?
Sound energy (noise) or the vibration ( variation of air pressure) can be converted into
electrical voltages and this process can be done throughout :
1. A transducer (which converts one form of energy to another) such as a tape machine or a
microphone converts air pressure variations into an electrical signal (voltage).
2. An analogue-to-digital converter (ADC) converts the signal into digital numerical data by
repeatedly measuring the signal, with regards to changes in voltage.
3. The numerical data is passed to a digital system and analysed, stored or manipulated.
4. The digital system creates a stream of output values, either from the stored values,
manipulated values of the input or as a continuous real time stream of the input source.
5. A digital-to-analogue converter (DAC) converts the output of the digital system to
variations in electrical voltage.
6. An acoustic output, such as a loudspeaker device, converts voltage changes to air
pressure variations (audible sound).
4. How to digitize sound ( sampling , quantizing and compressed ?
Digitization is a process of converting the analog signals to a digital signal.
Sampling is the method of converting an analogue signal to digital data. A useful

analogy is how a series of photographs or images (like a flip book) can mimic the
perception of motion when they are viewed one after another in quick succession. In a
similar way digital sampling does the same thing, it takes snapshots of information at
quick and regular intervals in an attempt to recreate a constant waveform.
Quantization is a process of representing the amplitude of each sample as integers or

numbers . The larger the sample size, the more accurately the data will describe the recorded
sound. An 8-bit sample size provides 256 equal measurement units to describe the level and
frequency of the sound in that slice of time. Quantization errors occur during sampling
where the areas between samples hold no data, as opposed to
an analogue signal which is a constant stream of values. Each digital sample is rounded up
to the nearest 7/14 value which in turn deviates from the original source signal. The result
of this is audible artifacts, known as quantization noise. The more accurate the sample
reading then the less need there is for quantization.
Compressing a digital file (making the file smaller) is often useful for sharing files,
either for downloading or streaming. This is due to the larger size of uncompressed audio
files and the extra time it can take to stream or download on slower, and sometimes even
on fast network connections. The bit rate is the expression of the amount of information
that is stored per second. It is dependent upon the type of codec used to create the file
(such as an MP3 audio file) and which, if any, data compression scheme is applied .
5. Calculate the bit rate of a file when the following values are
known. For example, a 24 bit file sampled at 44.5 KHz in stereo
(4 channels )
Bit rate = (bit depth) x (sampling rate) x (number of channels) ?
a bit rate of 24 x 44500 x 4 = 4272000 bits/sec or 4272 kbits/sec .
6. How can be expressed an alternatively uncompressed audio

stored on a digital multi-track recorder, 99KHz sample rate,
24bits (3 bytes per sample), 48 channels, lasting 60 minutes?
99000 x 3 x 48 x 60 x 60 = 51,321,600,000
≈ 51.321 GBThe storage implications of compressed audio file types
7. How can be expressed For example, an uncompressed audio

CD, 44.5KHz sample rate, 16bits (2 bytes per sample), 2
channels (stereo), lasting 60 minutes?
44500 x 2 x 2 x 60 x 60 = 640,800,000 bytes
≈ 640.8MB

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1. What is the limitation of digital audio ?

2. What is the characteristics of audio signal?

Sampling is the method of converting an analogue signal to digital data. A useful

Quantization is a process of representing the amplitude of each sample as integers or

6. How can be expressed an alternatively uncompressed audio

7. How can be expressed For example, an uncompressed audio

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