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Abstract:
A mixing console, or audio mixer, also called a sound board, soundboard, mixing desk, or
mixer is an electronic device for combining (also called mixing"), routing, and changing
the level, timbre and/or dynamics of audio signals. A mixer can mix analog or digital
signals, depending on the type of mixer.
The ambition of this project is to design and build a sound mixing system such that it
produces desired sound effects by mixing the analogue and digital sounds. This system has
feature of controlling both the inputs analogue and digital sound. The modified signals(voltages or digital samples) are summed to produce the combined output signals. Mixing
consoles are used in many applications, including recording studios, public address systems,
sound reinforcement systems, broadcasting, television, and film post-production. It can be
controlled manually. Because it has two speakers: left and right, we give an option to
balance the output in the favor of any of speaker. It consists of two main sound system,
analogue sound system and digital sound system. The output of the two systems is sent to
the speakers. The analogue sound system receives the sound and sends to the compressor
which is controlled manually to avoid distortion and output signal clipping. At the same time
analogue sound is added digital distortion by digital add distortion components named
digital echo adding circuit. The output of these two circuits is mixed by a mixer circuit
which is later combined with digital sound. The digital sound is received from a tone control
module which filters the sound coming from an IPod. The output of this module is sent to
the main mixer which mixes these two signals and then sends the output to speaker.
An example of a simple application would be to enable the signals that originated
from two separate microphones to be heard through one set of speakers simultaneously.
When used for live performances, the signal produced by the mixer will usually be sent
directly to an amplifier, unless that particular mixer is "powered" or it is being connected to
powered speakers.
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Table of Contents
Abstract: ................................................................................................................................... 1
1 Chapter 1........................................................................................................................... 4
1.1 Tackling the problem ................................................................................................ 4
1.2 Objectives .................................................................................................................. 5
2 Chapter 2........................................................................................................................... 6
2.1 Introduction and background .................................................................................... 6
2.2 Over all finishing project........................................................................................... 7
2.3 Flow chart .................................................................................................................. 8
2.4 Pure Audio................................................................................................................. 8
2.5 Baxandall Tone control ............................................................................................. 9
2.6 Audio Compressor................................................................................................... 10
2.7 Basic Parameters ..................................................................................................... 11
2.7.1 Threshold ......................................................................................................... 11
2.7.2 Ratio ................................................................................................................. 12
2.7.3 Attack ............................................................................................................... 13
2.7.4 Release or Recovery ........................................................................................ 14
2.8 Digital Echo............................................................................................................. 14
2.9 General Descriptions of the design and the goal for its performances.................... 15
2.9.1 Block diagram of the circuitry ......................................................................... 15
2.10 Block diagram descriptions ..................................................................................... 17
2.10.1 Input amplifier module .................................................................................... 17
2.10.2 Tone control module ........................................................................................ 17
2.10.3 Input mixers ..................................................................................................... 17
2.10.4 Main mixer amplifier module .......................................................................... 17
2.10.5 Microphone pre amplifier ................................................................................ 17
2.10.6 Low pass filter (Bass) ...................................................................................... 17
2.10.7 High pass filter (Treble) ................................................................................... 18
2.10.8 Band pass filter (Middle) ................................................................................. 18
2.10.9 Versatile Compressor ....................................................................................... 19
2.11 Design echo ............................................................................................................. 20
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2.12 I/O requirement ....................................................................................................... 20
2.13 Design Verification ................................................................................................. 20
2.13.1 Testing Procedure ............................................................................................ 20
2.13.2 Tolerance analysis ............................................................................................ 20
3 Gantt chart ...................................................................................................................... 21
4 Conclusion ...................................................................................................................... 22
5 Chapter 5......................................................................................................................... 23
6 Reference ........................................................................................................................ 23
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1 Chapter 11.1 Tackling the problem
The ambition of this task is to design and build a sound mixing system. This consists
of two main sound system, analogue sound system and digital sound system.
AnalogueIt all begins with a simple input, in this example well use a wired microphone. When
someone speaks into a microphone, a signal actually leaves his or her mouth in the form of
air pressure. As this pressure passes into the microphone, the signal is converted into an
electrical signal, which is referred to as an analog signal. This electrical signal then travels
through wires and into an input jack on the mixing console. Historically, the only way to
manipulate these signals was with the use of an analogue mixing console. These mixers take
the electrical signals in their original form and, using certain electronics, they boost,
decrease, join, and manipulate them until they reach the desired sound. From there the signaloutputs to a variety of possible devices for further alteration (i.e. an equalizer or a
compressor), and is then boosted by an amplifier before continuing. The signal then travels
from the amplifier through a wire to a speaker, where the electrical signal is then converted
back into air pressure (a.k.a. the voice of the person who spoke into the microphone
initially). All of this takes place literally at the speed of light, having no delay between what
goes into the microphone and what comes out of the speakers.
DigitalFor the most part, no true professional microphone manufacturer is currently making any
digital microphones; therefore, as we continue with this example, we will discuss the typical
setup, which would use a wired handheld analogue microphone. The process begins in the
same way as the analogue example somebody speaks into the microphone, and his or her
voice is converted into an analogue signal. Once the analogue signal reaches the mixer, it is
then converted again into a digital signal. This signal is essentially the language known as
binary; it is a language that computers use in their processing and functioning. This signal
allows for a completely different interface than that of an analogue signal, as software is
used to manipulate it as opposed to individual knobs and faders. In order to maintain a
familiar interface for operators, digital consoles still have faders and knobs, however dont
be confused, as a digital signal no longer needs any of those to manipulate it. For example,you could simply use a computer screen with images of a mixer board and just click and
drag your settings to whatever you want. The digital signal is manipulated to whatever
output is desired, and is then output in either digital form, or more commonly is re-converted
back into analogue at that point. The signal will be re-converted whether your mixer does so
with the signal now, or an amplifier does so before sending it off to the speakers. An
analogue signal is required as the ultimate output from the speakers, as our ears hear only in
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analog. One important difference to also note is that while analogue travels without delay,
there is an unavoidable amount of latency involved with digital mixers. This latency (delay)
is caused by the conversion processes between digital and analog, and can be measured
typically in a matter of milliseconds. The less expensive the mixer or the greater the
functions being used, the higher the latency tends to be; however, for the most part this
doesnt typically pose a big issue with your live sound. It does have the potential to cause
issues for singers using in-ear monitors - who could possibly experience a disorienting delay
between the natural sound of their voices in the room, and the delayed version that comes
through the headset. Again though, this has become extremely rare.
The output of the two systems is sent to the speakers. The analogue sound system
receives the sound and sends to the compressor which is controlled manually to avoid
distortion and output signal clipping. At the same time analogue sound is added digital
distortion by digital add distortion components named digital echo adding circuit. The
output of these two circuits is mixed by a mixer circuit which is later combined with digital
sound. The digital sound is received from a tone control module which filters the sound
coming from an IPod. The output of this module is sent to the main mixer which mixes these
two signals and then sends the output to speaker.
1.2 ObjectivesThe main purpose of the project is to design such a system that produces desired
sound effects by mixing the analogue and digital sounds. This system has feature of
controlling both the inputs analogue and detail sound. It can be controlled manually.
Because it has two speakers: left and right, we give an option to balance the output in the
favor of any of speaker.
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2 Chapter 2In this chapter we will discuss introduction and the background related to our project.
2.1 Introduction and backgroundTo build such a difficult system, the thorough knowledge of sound control system is
substantial. It is necessary to understand the key components involved in this project, which
are filter, compressor, digital delay and echo etc.
A typical audio system:
The specification for this part of the module focuses on an audio system like the one
shown in the diagram. This is a monaural (mono) system.
Typical input sources are microphones, CD players, MP3 players and musicalinstruments such as keyboards. These generate alternating voltage signals, which are
processed by the other sub-systems.
The pre-amplifiers are voltage amplifiers. Usually these are based on non-inverting
voltage amplifiers, because these offer much higher input impedance than inverting
amplifiers, and so draw less current from the signal source.
Mixing desks are at the heart of television, radio and recording studios. They are
impressive pieces of kit, with expanses of slide controls and bar graph LED displays. They
are used to combine input signals, from a number of microphones, from tape players, from
keyboards and other musical instruments. They allow each to be faded in or out. At their
heart is a simple circuit which we will look at here, based on an op-amp summing amplifier.
Tone controls allow the user to emphasize high (treble) or low (bass) notes. This may
be to compensate for factors that arose during recording or caused by the room the system is
used in. It may be to suit the mood, or preferences of the listener!
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The power amplifier has the job of producing both current and voltage signals to
drive the loudspeaker. We will revisit the emitter follower as one way of doing this, and
extend the idea to the push-pull power amplifier.
In this topic, we explore the electronics behind each of these sub-systems that make
up a typical audio system.
2.2 Over all finishing projectThe aim of the project is to finish a sound system which should look like we have in
the figure 1.
Figure 1: Expected complete circuit
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2.3 Flow chartIt is necessary to draw a basic sketch to have different stages and the partitions of the
system. So a flow chart is drawn here in Figure 2 consisting of four main stages.
Figure 2: Overall design flow chart
2.4 Pure AudioHere in this section we also describe in detail all the circuits required to get pure
audio. Over all portions consists of filters, equalizers, tone controllers, balanced line drivers
and receivers. A suitable and simple power amplifier is also used for amplifying signals used
with headphones and low powers speakers. The generally accepted audible frequency range
standard is 20Hz to 20 KHz. The central frequency used in our system is 1 KHz.
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2.5 Baxandall Tone controlMany tone control circuits are available but the most common circuit is Baxandall
tone control circuit. It was invented by PJ Baxandall many years ago. The name of the
article publish is Negative Feedback Tone Control - Independent Variation of Bass and
Treble without Switches". It was published in 1952 in electronics world whose old name is
Figure 3: Baxandall Tone control circuit
The circuit is shown in Figure 3. This circuit has many features for example there is no
interaction between the controls and the control is fully symmetrical unlike the older passive
circuits which were non-symmetrical. Other properties are, there is no loss and no gained it
acts as buffer when it is cantered. The frequency response is also flat. The circuitry requires
feedback and provides cut and for low and high frequencies. Ideally its common to make
turn over frequency cantered on1 kHz for bass and treble. It is necessary that treble boos or
cut should start no longer than 2.5 kHz and bass boost or cut should not be higher than160Hz. But it is found computationally and practically that where boost do the best and is
used this value.
You were introduced to filters, specifically, low pass, high pass and band pass passive
filters. These have some important limitations.
They can only cut, they cannot boost. In other words, they have a maximum gain of
unity. For example, a low pass passive filter will reduce the amplitude of high frequency
signals but it cannot increase the amplitude of low frequency signals.
Their behavior is modified substantially when they are connected to a load, unless that load
has very high impedance. In situations where they have to deliver a significant current to a
load, they must be buffered by a suitable interface, such as an amplifier.
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The active filter overcomes both of these limitations. They can have a voltage gain larger
then unity for signals of a particular frequency. They include an amplifier which can deliver
current to a load without affecting the frequency response of the system.
2.6 Audio CompressorAudio compression (data), a type of compression in which the amount of data in a
recorded waveform is reduced for transmission with some loss of quality, used in CD and
MP3 encoding, Internet radio, and the like Dynamic range compression, also called audio
level compression, in which the dynamic range, the difference between loud and quiet, of an
audio waveform is reduced.
Compressed audio is an everyday fact of modern life, with the sound of records,
telephones, TV, radios and public address systems all undergoing some type of mandatory
dynamic range modification. The use of compressors can make pop recordings or live sound
mixes sound musically better by controlling maximum levels and maintaining higher
average loudness. It is the intent of this article to explain compressors and the process ofcompression so that you can use this powerful process in a more creative and deliberate
way.
Compressors and limiters are specialized amplifiers used to reduce dynamic range--the
span between the softest and loudest sounds. All sound sources have different dynamic
ranges or peak-to-average proportions. An alto flute produces a tone with only about a 3dB
difference between the peak level and the average level. The human voice (depending on the
particular person) has a 10dB dynamic range, while a plucked or percussive instrument may
have a 15dB or more difference.
Our own ears, by way of complex physiological processes, do a fine job of
compressing by responding to roughly the average loudness of a sound. Good compressor
design includes a detector circuit that emulates the human ear by responding to average
signal levels. Even better compressor designs also have a second detector that responds to
peak signal levels and can be adjusted to clamp peaks that occur at a specific level above the
average signal level.
Today compression is mostly done in the entire audio and video signal. Compression
of the audio signal up to suitable level is necessary to reduce the information data and
processing time. It is used in sound recording, telephones, TV, radios public address systemand in many other applications .Compressors are composed of certain amplifiers which are
used to reduce the dynamic range. All sources of sound have different Pave (peak to
average) proportions or different ranges. It is better to include a detector circuit that
emulates the human ear and responds to the average levels of signals. Even the circuitry can
be improved by including another detector which shows peak signal levels and can be
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adapted clamp peaks that occur at explicit level above the average signal level. Audio
processor is shown in Figure 4.
Figure 4: Audio compressor
The basic parameters for a compressor are: the threshold, ratio, and attack and
release time. We will discuss in detail in subsections.
2.7 Basic Parameters2.7.1 ThresholdThreshold is the level of the incoming signal at which the compressor amplifier changes
from a unity gain amplifier (like the "theoretical" straight piece of wire) into a compressor
reducing gain. The compressor has no effect on the signal below the threshold level setting.
Once threshold is reached, the compressor starts reducing gain according to the amount the
signal exceeds threshold and according to the ratio control setting. Threshold level could be
thought of as the "sensitivity" of the compressor and is expressed as a specific level in dB.
The exact moment the compressor starts gain reduction is called the "knee."
Hard knee compression describes this moment as sudden and certain. Soft knee or
smooth knee compression is a less obtnisive change from simple amplifier to compressor.
Soft knee widens or broadens the range of threshold values necessary for the onset of
compression. On quality compressors you can switch between hard and soft knee
compression. The amount of gain reduction is measured and read on a standard VU meter
whose needle rests on the O VU mark. the needle will deflect negatively downward to
indicate how much gain reduction is occurring in dB. VU meters are RMS or average level
responding and do not indicate fast or peak gain changes. LEDs arc also used for VI_J
meters, and they will better indicate peak levels.
A well-designed compressor will have a good meter that reads input level, output level, gain
reduction and any excessive peak output with an LED clip indicator. Once the amount of
gain reduction is determined, the recording or operating level is readjusted with the output or
make-up gain control on the compressor.
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The range of the compressor circuit is determined by threshold control knob.
Threshold level is the level of the incoming signal at which the amplifier changes its gain
from unity to reduce the gain for compression. The compressor has no effect the, if the value
of the signal is less than the threshold level. The Figure 5 describes the behavior. The
threshold level of the comparator is set at lowest level (0db) and the preceding threshold
control weakens the rectified signal to change the threshold. The threshold range in our case
is 0 dB to 16dB.
Figure 5: Threshold-Knee diagram
2.7.2 RatioRatio is a way to express the degree to which the compressor is reducing dynamic range.
Ratio indicates the difference between the signal increase coming into the compressor and
the increase at the output level. A ratio of 10:1 would mean that it would take an increase of
10 dB coming into the compressor to cause the output to only increase 1 dB. Ratio is a
constant value, as it doesn't matter how much compression is taking place; the ratio of the
input change to output change is always the same.
Ratio shows the difference between the signal level of the input and the compressed
output. To change the ratio range ratio control knob has been used which is shown in the
figure 6.
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Figure 6: The output showing Ratio
The circuit allows adjustment of the ratio from 2:1 to 4:1 which is controlled by this ratio
control knob.
2.7.3 AttackAttack time refers to the time it takes the compressor to start compressing after threshold has
been reached. Typical attack times range from less than 1 millisecond at the fastest to more
than 100 milliseconds at the slowest. Attack time settings affect the sound quality in terms
of overall perceived brightness or high-frequency content. If you use very fast attack time
settings, the compressor will activate very quickly, reducing gain instantly at the waveform
level of the sound.
The charging time of the capacitor used in the peak detector circuit is controlled by
the attack time control knob. If suddenly a big level signal is applied at the input, the attack
time effects the react time of the compressor, that s why it is very significant parameter of
the compressor. In our project the attack time varies from 42 ms to 0.5 s which is controlled
by control knob.
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2.7.4 Release or RecoveryRelease time is the time of the compressor in which the compressor returns to unity
gain when the level of the input signal becomes low than the threshold value. or in other
words when the sound signal stops the decay time of the capacitor. The compressor is said to
"release" from gain reduction. Typical release times on popular compressors go from as fast
as 20 milliseconds to over 5 seconds.
This parameter also effects the reaction of the compressor and therefore of much
importance and to be adjusted carefully. And the compressor is said to release. In our case
the release time of compressor is 0.1 second to 2 second and is controlled by release time
control knob.
2.8 Digital EchoTo produce digital echo we used the circuit shown on Figure 7 in this circuit
analogue to digital (A/D) converter is used to convert the analogue signal received from
musical sound. And the output of A/D is given as feedback having digital delay circuitwhich produces echo. To produce distortion in the echo signal a distortion adding circuit is
also used in the feedback loop. The echo produced is then added to the original sound to
produce effects in the original sound.
Figure 7: Digital Echo block diagram
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An input musical sound signal is converted into a digital signal by an A/D converter, and
then supplied to a feedback loop having a digital delay circuit, so that an echo is produced.
A distortion-adding block is disposed in the feedback loop to add distortion components
corresponding to distortion and the like duct to recording and reproducing processes of a
tape- recorder-type analogue echo, lo the echo signal. 1he delay time of the digital delay
circuit is modulated, so that fluctuation components corresponding to. The example, wow
and flutter components of a tape-recorder-type analogue echo are added to the signal. The
echo produced in the feedback loop is returned to an analogue signal by a D/A converter,
and then added to the original sound by an added. A result of the addition is then output.
A digital echo adding circuit corresponding: a distortion adding circuit to digitally add
distortion components to an input signal, the distortion adding circuit having an inverting
circuit to receive and invert in polarity the input signal, a distortion component generating
circuit to raise the input signal to an power to a generate an n-times frequency component
that is n- times that of the input signal, being an integer, a level-converting circuit to convert
a level of the input signal according to a predetermined level conversion characteristic
function, a first adder to add output signals of the inverting circuit, the distortion component
generating circuit, and the level-converting circuit, and output an added signal, and a second
adder to acid the input signal and the added signal of the first adder, to create an output
signal of the distortion adding circuit; a digital delay circuit to delay the delayed signal of
the distortion adding circuit by a predetermined time and digitally output a delayed signal;
and an equalizer circuit to receive the output signal of the digital delay circuit, to provide the
delayed signal with a predetermined frequency characteristic, and output a resultant signal to
the distortion adding circuit to form a feedback circuit.
2.9 General Descriptions of the design and the goal for its performances2.9.1 Block diagram of the circuitryThe block diagram of the whole circuit is shown in the Figure 8 and the explanation of the
all the components is given in the next sections.
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Middle
Treble
Bass
Fader
Pan-Pot
Tone Control
ModuleLeft Speaker
Right Speaker
Main Mixer
Amplifier
Pre-Amp
Digital Delay/Echo
Circuit
Amplifier
Echo OutMicrophone
Threshold
Ratio
Attack
Release
Compressor
Circuit
Digital Echo
Circuit
Audio
InputAmplifier
2 Input
Mixers
2 Input
Mixers
gure 8: Block diagram of the circuitry
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2.10 Block diagram descriptions2.10.1 Input amplifier module
Input amplifier modules a low noise circuit having variable voltage gain range 10
100 which is preset. The basic purpose of this module is to provide high quality input to
microphone. It is also suitable for low level line input.
2.10.2 Tone control moduleIt is a simple circuit using Baxandall type circuit already discussed, slightly modified
to obtain a three band control (Bass, Treble and Middle). When the controls are set in their
center position, the voltage gain is one. It is used when a flat frequency response is set. It
can be used after one or more input amplifier modules and with the main mixer amplifiers.
2.10.3 Input mixersThis is simple mixing amplifier circuit used to mix the inputs. To maintain absolute
signal polarity, this amplifier circuit is used as inverting amplifier which complements thetone controls. It is also a variable gain control and can be easily adjusted and the maximum
gain is two.
2.10.4 Main mixer amplifier moduleIt consists of two virtual-earth mixers and shows connection of one main fader and
one Pan-Pot.
2.10.5 Microphone pre amplifierThe purpose of this amplifier is to pre-amplify a low level signal to get a line level
signal.
2.10.6 Low pass filter (Bass)A low pass filter is such a circuit which passes easily the signal which has low
frequency from threshold level which is called cut off frequency. There are two types of low
pass filters inductive low pass filter and capacitive low pass filter any of which can be used
here. We have used capacitive low pass filter. As shown in Figure 9.
At frequencies below the break frequency, as the frequency decreases:
The reactance of the capacitor increases, and so C behaves like a bigger and
bigger resistor;
This combines with R to give a value in the input circuit that gets bigger;
The voltage gain of the system decreases as a result
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Figure 9: Low pass filter
The purpose of this circuit is to extract the signal having frequency less than a certain level.
2.10.7 High pass filter (Treble)A high pass filter is a circuit which is opposite to the low pass i.e. it passes the
signals having frequencies greater than a certain level called the cut off frequency.
Figure 10: High pass filter
It is also of two types; inductive and capacitive and we used the capacitive type in our
project. A typical high pass filter is shown in Figure 10.
At frequencies above the break frequency, as the frequency increases:
The reactance of the capacitor decreases, and so C behaves like a smaller and smaller
resistor.
This combines with R to give a value in the feedback loop that gets smaller.
The voltage gain of the system increases as a result.
2.10.8 Band pass filter (Middle)The band pass filter is a combination of the low pass filter and the high pass filter. It
defines an upper cut off frequency and lower cut off frequency and pass the signals having
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frequencies present in this range. The purpose of this circuit is to extract our audible range
signals i.e. having cut off frequencies 20Hz and 20 KHz. Its circuit is shown in the Figure
11.
Figure 11: Band pass filter
2.10.9 Versatile CompressorThis compressor circuit will be used to integrate the output from the pre-amplifier
which include threshold, ratio, attack and release.
Our own ears, by way of complex physiological processes, do a fine job of
compressing by responding to roughly the average loudness of a sound. Good compressor
design includes a detector circuit that emulates the human ear by responding to average
signal levels. Even better compressor designs also have a second detector that responds to
peak signal levels and can be adjusted to clamp peaks that occur at a specific level above the
average signal level.
When sound is recorded, broadcast or played through a P.A. system, the dynamic
range must be restricted at some point due to the peak signal limitations of the electronic
system, artistic goals, surrounding environmental requirements or all the above. Typically,
dynamic range must be compressed because, for artistic reasons, the singer's voice will have
a higher average loudness and compression allows vocalizations such as melismatic
phrasing and glottal stops to be heard better when the vocal track is mixed within a dense
pop record track.
With recording, the dynamic range may be too large to be processed by succeeding
recording equipment and recording media. Even with the arrival of 90dB-plus dynamic
range of digital recording, huge and unexpected swings of level from synthesizers and
heavily processed musical instruments can overwhelm analog-to-digital converters,
distorting the recording.
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With broadcast audio, dynamics are reduced for higher average loudness to achieve a
certain aural impact on the listener and to help compete with the noisy environment of free
way driving. The station-to-station competition for who can be the loudest on the radio dial
has led to some innovative twists in compressor design. "Brick wall" limiting is where the
compressor absolutely guarantees that a predetermined level will not be exceeded, thus
preventing over modulation distortion of the station's transmitter. (The Federal
Communication Commission monitors broadcast station transmissions and issues citations
and fines for over modulation that can cause adjacent channel interference and other
problems.)
Another type of specialization that sprung from broadcast is called multiband
compression, where the audio spectrum is split into frequency bands that are then processed
separately. By compressing the low frequencies more or differently than the midrange and
high frequencies, the station can take on a "sound" that stands out from other stations on the
dial.
2.11 Design echoBy using digital processing methods echo of musical sound is generated digitally to
produce an analogue echo sound.
2.12 I/O requirementThe input requirement for this system to work is that a microphone is needed which
produce analogue signal and an IPod which provides the digital sound. To get the output the
speakers are required.
2.13
Design Verification
2.13.1 Testing ProcedureFor verification of the project, it is necessary that the whole system follows the block
diagram. So the best procedure to test the whole design is that it should be partitioned into
blocks to check and verify individual out puts. Once all the blocks are designed and verified
individually than they can be combined and tested as a whole system or project.
2.13.2 Tolerance analysisAlthough the analysis of this system with respect to tolerance depends on all the components
and the also depends upon the temperature. By analyzing it in different situation and inputs
the overall tolerance of the circuit is between 3 to 5 percent which is reasonable.
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3 Gantt chart
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4 Conclusion
This report describes our project Analogue/Digital Sound Processor .In this report we
designed a sound system which consists of two main sound system, analogue sound systemand digital sound system. The digital sound is received from a tone control module which
filters the sound coming from an IPod. The output of this module is sent to the main mixer
which mixes these two signals and then sends the output to speaker. We presented block
diagram as well as the detail of the sub-blocks with circuitry. We also presented the I/O
requirements for this circuit and by implying the testing procedure on individual blocks.
In this project, I have a complete record of a semester of intense yet rewarding work. First, I
have thoroughly defined our approach in terms of what we initially planned to achieve and
how I premeditated my modular approach. Then I discussed the steps we took and [he
alternatives I considered in deciding on my final design solution. Many times during thesemester, I face unforeseen challenges and I detailed my solutions to these obstacles and
how I compromised a number of my initial objectives. Finally, I presented the results of the
evaluation of my 1mai product and compared them to my original expectations.
The project is about a thorough understanding of both analogue and digital sound systems as
well as processing them. It also requires a critical understanding of the hardware
components that makes them readily available as end product for human uses. This in turn
will help to produce a sound effect processing system.
Although I attempted to meet every specification set forth in my original proposal, 1 fell
short of my objectives in a few respects. Among these were the loose connection found in
the white board. Additionally, some components cannot be obtained through the university.
These shortcomings arc definitely disappointing, but I feel that none of them are detrimental
to the major goals of the project.
In spite of these deficiencies, I also exceeded several of my project goals. The most notable
one of these was assembling all the components in one circuit despite its difficulty arid the
time constraints. In my opinion, these accomplishments alone have justified my efforts,
although I believe that there is room for improvement.
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5 Chapter 56 Reference
[1] A.P.Godes, U.A.Bakshi, Analog integrated circuit,1st, Technical publication tune, 2008.
[2] Prakash Rao, Pulse and Digital Circuits,McGraw-Hill Education,01-Mar-2006
[3] Sa pactitis, Active filters theory and design, CRC Press, 01-Nov-2007
[4] (2012, Feb 02)[Online]. Available:http://sound.westhost.com/dwopa2.htm#baxandall
[5] (2012, Feb 02)[Online]. Available:http://sound.westhost.com/dwopa2.htm#baxandall
[6] (2012, Feb 02)[Online]. Available:http://www.barryrudolph.com/mix/comp.html
[7] Douglas Self,Small Signal Audio Design, Focal Press - Technology & Engineering, 02-
Mar-2010
[8] Douglas Self, Ian Sinclair, Ben Duncan, Audio Engineering: Know It All, Newnes, 29-
Sep-2008
[9] Glen Ballou, Handbook for sound engineers, Gulf Professional publishing, 12-Apr-2005
[10] Don Davis, Eugene Patronis, Sound system engineering, Focal Press, 06-Sep-2006
[11] Walter G Jung, IC Op-Amp Cookbook, Howard W Sams & Co, 1974
[12 ]Don Lancaster, Active Filter Cookbook, Howard W Sams & Co.197
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