ANALOG &

DIGITAL SOUND.

B . GHOSH
Dy.Director (Engg)
NABM-BBSR
 SOUND PRESSURE AND PRESSURE
LEVELpoint
The sound pressure at a certain : is the difference between
the instantaneous pressure and the ambient mean pressure.
The unit of Sound Pressure (p) is Pascal (Pa) which is equal to
Newton per Square Meter (N/ m2).
Sound Pressure Level (SPL) is the optimum value of the sound
pressure for a particular environment ,condition or a device.
The reference value for Sound Pressure Level (SPL) is
twenty micro-Pascal (20 µPa = 20 x 10 - 6 Pa). From this SPL of
a 1 KHz sound a normal human ear starts hearing. This
reference level is called “Threshold – of – Hearing” for a young
person with normal hearing ability.
The Sound Pressure Level (Lp) is defined by the formula :

A pressure equal to the reference value is thus equal to

zero dB while 1 Pa equals 94 dB (93.98 dB) (Loud classical
music or heavy traffic.)
 The SPL value at 120 dB is called “Threshold – of – Pain”
LOUDNESS & PHON

Fletcher-Munson
Curves for
Equal loudness
 Loudness is defined as the intensity of sound as judged by the ear.
 It needs higher intensity at low frequencies than at high
frequencies to impart same sensation of loudness.
 The intensity of 60 dB at 40 Hz and of 0 dB at 1000 Hz imparts
the same loudness.
 The intensity in dB with reference to Threshold of hearing as
perceived by the ear at 1000 Hz is called phon (P).
 If it is 0 dB then loudness is 0 phon, if it is 40 dB then loudness is
40 phon.
Intelligibility- It is defined as the clearness of one’s speech
determined through the test of articulation.
The person under test is made to speak syllables in random order,
which is recorded and heard by a group of persons.
The articulation efficiency should be about 90% for broadcast
purpose & 80% for telephone speech.
Overtones and Timbre: Sound waves produced by Speech and
musical instruments are not pure sine waves, but are complex
waves consisting not only the fundamental frequencies (tones) but
also of their harmonics, and other frequencies, called ‘overtones’.
The proportion of tones & overtones present in the sound that helps
us to identify any particular voice is called Timbre.
 FREQUENCY RESPONSE
The frequency response of a device describes the relationship
between the device's input and output with regard to signal
frequency and amplitude.

Plotted frequency response of the DUT (Relative level vs. Frequency).

The frequency response describes the usable range of signal

frequencies which the device will pass from input to output.
 The frequency response gives an indication of the fidelity
with which the device transfers a signal from input to output.

 The less deviation there is, in output level across the stated
frequency band, the more faithfully the signal at the output will
reflect the signal at the input.
 SPECIFICATION IN FREQUENCY RESPONSE
FREQUENCY RESPONSE: 30 Hz to 18 kHz, ±3 dB : The range
(from 30 Hz to 18 kHz) is accompanied by the qualifier,
«+ or - 3 dB." This is called the tolerance of the specification.
The tolerance tells us the maximum deviation in output level that
we can expect over the stated range if the input level remains the
same at all frequencies.

frequency response "flat" frequency response "flat" frequency response(WB)

(20 Hz to 20 kHz, +0, -1 dB. ) (10 Hz to 40 kHz, +0, -3 dB.)

Without a stated tolerance, the frequency response specification is

useless, since we are left to guess the unit's effect on the signal.
There might, in fact, be horrendous peaks and/or dips in the
response - and these could alter the signal considerably.
 VOLUME, LEVEL AND GAIN
Three of the most often misused or carelessly used terms in audio are
Volume, Level, And Gain.
VOLUME :
It is defined as power level. of the audio equipment. Turning up the
volume, means increasing the power.
Unfortunately, the term is loosely used to describe sound intensity or
the magnitude of an electrical signal.

LEVEL :
Level is defined as the magnitude of a quantity in relation to an
arbitrary reference value , Sound Pressure Level (SPL), for example,
expresses level in Db.

The audio level in a signal processor, for example, may be expressed

in dBm, which is referenced to 1 milli-watt.

GAIN :
Gain is usually assumed to be the input to output parameter
Increase/decrease of a signal, usually expressed in dB.
 DYNAMIC RANGE & HEADROOM

oThe difference, in decibels, between the loudest and the quietest

portion of a program is known as its dynamic range.
oIf program quietest part is obscured by ambient noise, the dynamic
range is the diff. between program loudest part & noise floor.
oThe inherent noise floor of audio broadcasting ,is the residual
electronic noise in the system.
 DYNAMIC RANGE OF A TYPICAL ROCK CONCERT
Rock/Music -concert has the widest dynamic range you're ever
likely to encounter.
The sound levels at the microphones (not in the audience) may
range from 40 dB SPL (the audience, wind, and traffic noise at the
mic during a very quiet, momentary pause) to 130 dB SPL .
 The dynamic range of the concert is obtained by subtracting the
noise floor from the peak levels:
Dynamic Range = (Peak Level) - (Noise Floor)
= 130 dB SPL - 40 dB SPL = 90dB.
The concert has a 90 dB dynamic range at the microphone.
 The electrical signal level in the sound equipment is proportional to
the original sound pressure level (in dB SPL) at the microphone.
When the sound levels reach 130 dB SPL at the mic, the maximum
line levels (at the mixing console) may highest, and maximum
output levels from each power amplifier may peak at full power.
Similarly, when the sound level falls to 40 dB SPL, the minimum
line level falls drastically and power amplifier output level falls to
250 nano-watts.
 The speakers should be capable of this range, or they may either
distort or burn out on the peaks.
HEADROOM :
The electronic line level in the concert sound system corresponds to
an average sound level of 110 dB SPL at the microphone called the
nominal program level.

The difference between the nominal and the highest (peak) levels in
a program is the headroom. In this case the headroom is :
Headroom = (Peak Level) - (Nominal Level)
= 130 dB SPL - 110 dB SPL =20dB.

•The Signal-To-Noise Ratio, represents the difference between

the nominal level and the noise floor.
SOUND SYSTEM INADEQUATE FOR A PARTICULAR DYNAMIC RANGE & SOLUTION

When the dynamic range of the program material exceeds the

dynamic range capability of the sound system, some
combination of the following will result:
a) Program peaks will be distorted due to clipping and/or
loudspeaker break-up.
b) Quiet passages will not be heard because they will be below
the electrical and/or acoustic noise floor.
In the above case the dynamic range of the acoustic level to
the microphone, the mixing console , the power amplifier, & the
speaker system all should maintain the value of 90 dB .
Suppose during a particular OB coverage the mixing console
gets damaged and you need to hire a console from a local
sound system shop.
• We measure the rented electronics and find they have an electronic
noise floor of -56 dBu (1.23 mill volts), and a peak output level of +18
dBu (6.16 volts) , that is a dynamic range of 74 dB, a drastic fall of
16 dB.
 Compression of a 90 dB program. Compression above threshold.
The program still has an acoustic dynamic range of 90 dB , as
shown in Figure above.
 But as a result of sub-quality mixing console there may be extreme
clipping of program peaks, where the console output cannot rise high
enough in level to follow the highest program levels.
 Quiet passages, corresponding to the lowest signal levels, may be
buried in the noise floor.
FITTING WIDE PROGRAM DYNAMICS INTO LIMITED DYNAMIC RANGE
We saw that in ideal case the 90 dB dynamic range has
been maintained from microphone to the mixing console , the
power amplifier, & the speaker system ultimately .For every
2 dB change of input level, the output level changes by 2 dB.
Suppose for every 2 dB change of input level, the output
would change only 1 dB. What would happen to the dynamic
range of the program? It would be cut in half, 90 dB 45 dB.
This can be done with a simple signal processing device
known as a compressor , by setting its compression ratio of2:1,
making every dB of input level change into a half a dB of
output level change.
To get the program dynamics from 90 dB down to 74 dB ,
that for a 16 dB reduction in dynamic range. It is possible to
set the compressor for a 1.21 : 1 compression ratio, which would
squeeze 90 dB down to 74 dB.
Sometimes compression in the lower side makes breathing
sound louder, creating a pumping effect and low frequency
distortion . This can be handles by Threshold compression.
 DIGITAL AUDIO
Analog recording is done by converting continuous variations in
sound pressure into electrical voltage of continuous variations.
The sound source may be line input or from a microphone. The
varying voltage is then converted into a varying pattern of
magnetization on a tape for preservation and playback.
Analog replay system cannot differentiate between wanted &
unwanted signals like distortions, noise etc of the recording process.
 Recording process and playback system Imperfections are
reproduced as clicks, crackles & noises degrading final sound quality.
 Where as Digital recording converts the electrical waveform of
sound into a series of binary numbers, each representing the
amplitude of the signal at a unique point in time. These numbers are
stored in a coded form and the integrity of the signal is retained.
 The reproducing device is smart enough to distinguish between the
genuine and unwanted signals introduced during the conversion
process and reject them and keep wanted information almost intact.
Here the quality remains unaffected by distortions & imperfections
in the storage/transmission process within the system design limits,
through systematic correction process of timing and data errors.
 PULSE CODE MODULATION (PCM)

The device used to convert analog signal to digital signal is called

Analogue-to-Digital convertor (ADC).The reconversion process is
done by Digital-to-Analogue convertor (DAC).

Pulse Code Modulation (PCM) is a universally accepted process for

digitization of audio & video analogue signal. Pulse Code Modulation
involves three basic steps:
Sampling – PAM.
Quantization.
Encoding

Basic steps of pulse code modulation

Sampling (PAM): The sampling process means the taking of a number
of sample values at evenly spaced intervals across a continuous
signal.

Amplitude
Amplitude

Time Time

Period Period

Sampling Process with different Sampling Periods

The time lapse between successive sampling instants is called sampling
period. The sampling rate (number of samples per second) and period of a
system determines its overall bandwidth.
 A system with higher sample rates is capable of storing more frequencies at
its upper limit.
Sub-sampling or insufficient number of samples results in poor assessment
and possible misconception of the original form of the sampled signal.

Similarly, oversampling or too-many samples can lead to an unnecessary

accumulation of redundant data. But in many cases oversampling is desirable.
 ALIASING EFFECT:

Aliasing due to overlapping of side-bands No Aliasing

If the base-band is supplied having an excessive bandwidth, that is
more than half the sampling rate ‘ fs’ in use, the sidebands will
overlap, resulting in is what is called aliasing.
By definition an ‘alias’ is an undesired depiction of the original
signal arising due to sub-sampling, when the sampled signal is
reconstructed during digital-to-analog conversion.
Here ‘ fn’ the Nyquist cut-off frequency is the maximum permissible
base-band frequency, equal to half the sampling frequency fs.
Aliasing does not occur when the input bandwidth is equal to or less
than half the sampling rate fs., that is “the sampling rate must be at
least twice the input bandwidth or the highest frequency in the
baseband signal”.
 QUANTIZATION:

Quantizing is the process of interpreting the value of the amplitude

of each PAM sample with a set of predetermined equidistant levels
(steps) by approximating to the nearest predetermined level.
The number of levels that can be expressed as 2n (2 to the power of n),
where “n” is the selected number of bits.
A sample whose amplitude falls anywhere within a particular
interval is represented by a single value (quantized value) that of the
mid-point of the interval.
The reconstructed sample is a close approximation of the sampled
signal. The discrepancy between the actual signal and reconstructed
one is minimized by increases the number of quantization levels.
 QUANTIZATION ERROR
Between the quantizer discrete levels the sample value will be
rounded to the nearest mid-point of that interval leading to
Quantization Noise (Error) with R.M.S. amplitude of (Q =Interval) .
The magnitude of the quantizing error will be a maximum of plus or
minus half the amplitude of one quantizing step (Q).
A greater number of bits will result in a smaller error. But this will
result in increasing the bandwidth, or reducing the number of
multiplexed channels. An 8 bit offers 256 steps, 16 bit 65,536 steps .
Three bit Quantization

Quantized Output Levels

Four bit Quantization
 SIGNAL TO NOISE RATIO, DITHER & JITTER IN DIGITAL AUDIO.

Dynamic range Dithering

The signal-to-noise ratio of an

ideal n bit quantized audio S/N as a linear function of
bits & quantizing steps.
signal is :
= [6.02n + 1.76 ] dB.
Dither
The A/D converter is coarse at low levels signal samples & cannot make a
decision of the nature of the waveform. The resulting conversion is a falsely
converted into a square wave.
 All other slightly lower signals than the above are discarded giving rise to
distortion of information in the lower level area called “quantizing distortion”.
The affects of quantizing distortion can be eliminated by the addition of
some deliberately added random noise called Dither.
Jitter : Jitter refers to clock speed or sample timing variations in digital
systems. In audio it gives rise to effects of a similar technical nature to wow &
flutter . (Audio pitch variations due to speed variations in the analog systems).
That's All For Today ,

“THANKS”
B. GHOSH
DD(E)
NABM-BBSR
The sampling rate choice for audio of 44.1 KHz is based upon the field rate
and field structure of the existing television standards “525 lines at 60 Hz”
and “625 lines at 50 Hz” so that the samples can be stored on each usable
TV line in the field. Thus 44.1 KHz, a common multiple of both, and also
suitable for use as a sampling rate was chosen.
In 525 lines/ 60 Hz video, there are 35 blanked lines, leaving 490 lines
per frame or 245 lines per field. If 3 samples are stored per line, the
sampling rate given by:
60 x 245 x 3 = 44.1 kHz.

In 625 lines/50 Hz video, there are 37 lines of blanking, leaving 588

active lines per frame, or 294 per field, so the sampling rate is given by:
50 x 294 x 3 = 44.1 kHz.
DVD
DVD was the natural successor to CD, being a higher density optical disc format aimed at
the consumer market, having the same diameter as CD and many similar physical
features. It uses a different laser wavelength to CD (635 – 650 nm as opposed to 780 nm)
so multi-standard drives need to be able to accommodate both. Data storage capacity
depends on the number of sides and layers to the disc, but ranges from 4.7 Gbytes
(single-layer, single-sided) up to about 18 Gbytes (double-layer, double-sided). The data
transfer rate at ‘ one times ’ speed is just over 11 Mbit/s.
DVD can be used as a general purpose data storage medium. Like CD, there are
numerous different variants on the recordable DVD, partly owing to competition
between the numerous different ‘ factions ’ in the DVD consortium. These include DVD-
R, DVD-RAM, DVD-RW and DVD RW, all of which are based on similar principles but have
slightly different features, leading to a compatibility minefi eld (see Fact File 10.2 ).
Blu-Ray disk
The Blu-Ray disk is a higher density optical disk format than DVD, which uses a shorter
wavelength blue-violet laser (wavelength 405 nm) to achieve a high packing density of
data on the disk surface. Single-layer disks offer 25 Gbytes of storage and dual-layer
disks offer 50 Gbytes, and the basic transfer rate is also higher than DVD at around 36
Mbit/s although a higher rate of 54 Mbit/s is required for HD movie replay, which is
achieved by using at least 1.5 times playback speed.