(final 2)preliminary report

7/31/2019 (Final 2)Preliminary Report

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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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(final 2)preliminary report

Documents