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Secondly, there is the force exerted on the movable part by its suspension and by air sealed within the speaker enclosure, which combine to create the stiffness of the support for the movable part. These produce a reaction, or restoring, force which acts in a direction to maintain the movable part in its neutral position and this reaction force is equal in magnitude and opposite in direction to the distance of the movable part from its neutral position.

Systems employing an integrator or differentiator are based on a recognition that the acceleration of the movable part is proportional to the time derivative of its velocity, while the excursion of the movable part is proportional to the time integral of its velocity.

While, in theory therefore, an integrator or differentiator connected in feedback with an amplifier should help significantly to compensate for one source or another of loudspeaker distortion, it has been found in practice that prior art arrangements of this type are of limited utility. One reason for this is that it is difficult to properly adjust such devices so as to obtain accurate feedback signals of appropriate amplitude for a given speaker.

It has also been found that such a system functions properly only when the relative amplitude of the signal components provided by, or the gain of, the integrator or differentiator is kept small, so that the system will only provide satisfactory correction for small amounts of speaker distortion. In fact, further, if the gain of the integrator or differentiator of such a circuit is increased above a very low level, toward the value required to completely correct for one source of distortion in a speaker of average quality, it tends, because of the feedback connection, to generate instabilities which themselves measurably distort the resulting sound reproduction.

It is believed that one reason for these shortcomings is that such prior art systems are designed in dependence on the assumptions that the restoring force acting on the movable part of a speaker is linearly proportional to the excursion of that part from its neutral position and that the effective mass of the movable part of the speaker is constant, which assumptions are only first order approximations of the conditions found to exist in practice.

In systems employing a differentiator or integrator in feedback, and designed according to the abovementioned assumptions, the differences between the approximations which those assumptions represent and the relationships which exist in practice themselves cause signal reproduction errors whose amplitudes bear a positive exponential relation to such differences, unless the gain of the integrator or differentiator is turned down to such a low level that the element no longer has a significant beneficial influence on the resulting sound reproduction.

It is a primary object of the present invention to eliminate the drawbacks of such prior art arrangements. Another object of the invention is to substantially improve the sound reproduction quality of any loudspeaker. A more specific object of the invention is to supply such a loudspeaker with a signal made up of components whose relative amplitudes can be easily adjusted to compensate for major sources of distortion in any existing loudspeaker.

A further specific object of the invention is to achieve such distortion reduction through the use of integrators and differentiators connected in such a manner as to avoid the drawbacks presented by prior art systems utilizing such devices. These and other objects according to the invention are achieved by a novel electrical circuit unit arranged to be connected between the source of a time varying electrical signal and a transducer, such as a loudspeaker, having an electrical signal input connection, to cause the mechanical output produced by the transducer to constitute an accurate representation of the electrical signal.

The circuit according to the invention essentially includes an input terminal for receiving the electrical signal from the source, an output terminal arranged to be connected to the transducer signal input, a signal transmission path connected between the input terminal and the output terminal for supplying to the output terminal a signal component linearly proportional to the electrical signal, and a distortion compensating member having an input connected to the input terminal and an output connected to the output terminal for producing, at its output, a signal component constituting a direct function of at least one of the time derivative and the time integral of the signal appearing at its input.

Preferably, a circuit according to the present invention includes both an integrator and a differentiator, the gain of each being individually adjustable in dependence on the particular characteristics of the speaker, or speakers, to be driven by the circuit output signal. The invention is based on applicant's discovery that the connecting at least one of an integrator and differentiator in parallel with a signal component proportional to the original electrical signal, with the integrator and differentiator being connected to conduct in the forward direction of that signal component, significant compensation for sources of distortion in the responsible loudspeaker can be achieved despite the errors in the assumptions that the restoring force on the movable speaker part is linearly proportional to the excursion thereof and that the effective mass of the movable part is constant.

Moreover, in a circuit according to the invention, the integrator or differentiator can be set to have any desired gain without any danger of the generation of instabilities. It is thus possible to individually adjust the gain of an integrator and a differentiator to effect significant compensation for the sources of distortion produced by the inertia of the movable part and the stiffness of its suspension.

The integrator and differentiator can be adjusted independently without the adjustment of one having any influence on the contribution of the other and this substantially simplifies the adjustment of a system according to the invention to any particular speaker. The above-described circuit according to the invention achieves effective distortion reduction, even for a speaker whose actual operating characteristics differ from the assumptions underlying the use of integrators and differentiators.

In contrast to a feedback system, the forward connection of the integrator and differentiator according to the invention offers the advantages that the magnitude of the resulting departure from complete distortion reduction will simply be linearly proportional to the difference between the actual and assumed speaker characteristics, while in a feedback system, such difference produces a distortion correction error bearing a quadratic, or exponential relation to the difference. Moreover, a forward-connected system according to the invention is not subject to the instabilities associated with a feedback system.

A further development according to the invention permits compensation to be made for the fact that the restoring force acting on a speaker is not precisely proportional to its excursion and that the effective mass of the movable part of a speaker is not constant. Applicant has observed that the stiffness of the movable part of the speaker is often more nearly proportional to some power greater than 1 of the excursion of the movable part and that the inertial force produced by the movable part is similarly often proportional to some power greater than 1 of the acceleration.

The exponent of the power in each case depends to a certain extent on the particular speaker so that in order to provide a system which can be adjusted to a given speaker, embodiments of the invention employ suitable function generators at the output of the integrator and differentiator, these generators being of a type which can be adjusted to produce an output whose amplitude is a selected power, greater than or equal to 1, of the signal at its input.

If each such function generator is adjustable so that its output can be varied between a value corresponding to the first power of the signal at its input and a value corresponding to the fourth power of the signal at its input, the resulting system can be adjusted to provide nearly complete compensation for the distortions produced by the mass and stiffness present any existing loudspeaker sold for high fidelity systems, this including even relatively inexpensive, and thus low quality, speakers.

In embodiments of the invention, use can also be made of function generators which are designed to produce some other type of function. Signal source 1 can be constituted by any desired type of device, such as a microphone, record or tape player, or broadcast receiver. Power amplifier 2, which is here shown as a single block, can be constituted by any desired type of sound amplifying system, and thus may be constituted by any required number of stages housed in any number of units constituting, for example, a pre-amplifier and a main amplifier.

In accordance with the present invention, there is connected between the signal source and the power amplifier, a loudspeaker distortion reduction device 4 which, in preferred embodiments, includes a linear gain unit 12, an integrator 13 and a differentiator 14, all three of these components being connected in parallel to conduct signals in the same direction and having their outputs connected to a suitable mixer which algebraically combines the outputs of the unit 12, integrator 13 and differentiator 14 to produce a loudspeaker distortion compensating signal which is supplied to the input of power amplifier 2.

If power amplifier 2 is a multi-stage amplifier, the loudspeaker distortion reduction device 4 can, in further accordance with the invention, be connected between any desired pair of stages. The only real limitation in this regard is that device 4 be connected at a point in the amplifying chain where the signal power levels correspond to the capabilities of the component parts of the distortion reduction device, although it is understood that embodiments of such a device can be designed to operate at any desired power level.

A device according to the invention could even be connected between the output of the last power amplifier and the speaker input. In fact, one attractive possibility would be for a circuit according to the invention to be built into a speaker and to the preset to the speaker characteristics at the factory.

In this way, the performance of a speaker could be significantly upgraded at a very small additional cost. The integrator 13 and differentiator 14 are preferably constructed to each have an adjustable gain to enable the proportional contribution of each of these components to the total signal supplied to the speaker to be adjusted to the characteristics of the particular speaker employed in the system. In this way, a given embodiment of the invention can be readily adjusted, by means of only two manual controls, to any one of a wide range of loudspeakers.

Adjustment to the particular speaker connected to the power amplifier is quite simple, particularly since adjustment of either one of the components 13 and 14 will not alter the component provided by the other. Referring now to FIG. The values of the passive components employed in the actual circuit are indicated. The linear gain unit of this circuit is composed simply of a series resistor 23 and a shunt-connected adjustable resistor 24 which can be adjusted to vary the proportional contribution of the linear gain unit to the total signal supplied to mixer It will be observed that the integrator 13 and the differentiator 14 are relatively simple circuit devices each constructed according to principles well known in the art.

One of the principal advantages of the present invention is that it can be embodied by relatively simple circuits and can thus be fabricated at low cost. Integrator 13 includes a first high gain differential amplifier 31 having its negative input connected to the loudspeaker distortion reduction device signal input via a series resistor Connected in feedback to the negative input of the amplifier is a parallel arrangement of a further resistor 33 and a capacitor The positive input terminal of amplifier 31 is connected to ground.

The output of this amplifier is connected across a first potentiometer 35 whose movable tap is connected to the positive input of a second high gain differential amplifier 36 having a direct feedback connection to its negative input terminal. The movable tap of potentiometer 35 can be set to vary the proportional contribution, or gain, of integrator 13 to the total signal produced by the distortion reduction device.

In view of the connection of the system input to the negative terminal of amplifier 31, the output from the amplifier 36 will be a negative representation of the time integral of the signal applied to the system input. The differentiator 14 includes a first high gain differential amplifier 41 having its negative input connected to the system signal input via a series arrangement of a capacitor 42 and a resistor The negative input terminal of amplifier 41 is also connected in feedback to its output via a further resistor 44, while the positive input of amplifier 41 is connected to ground.

The output of amplifier 41 is connected to a potentiometer 45 whose movable tap is connected to the positive input of a second high gain differential amplifier The negative input terminal of this latter amplifier is provided with a direct feedback connection from the amplifier output.

Here again, the output of differentiator 14 is the negative time derivative of the signal supplied to the system input terminal. In mixer 15, the outputs from integrator 13 and differentiator 14 are supplied to the negative input terminal of a high gain differential amplifier 51 via respective coupling resistors 52 and The negative input terminal of amplifier 51 is also connected in feedback with its output via a resistor Finally, the output of linear gain unit 12 is connected directly to the positive input terminal of amplifier 51, and the output of the distortion reduction device is provided by the output of amplifier 51, which is connected to a suitable external output terminal.

The connection of the outputs of integrator 13 and differentiator 14 to the negative input of amplifier 51 cancels out the polarity reversal occurring in the integrator and differentiator, and thus restores the desired polarity relation between the integrated and differentiated components, on the one hand, and the component, provided by the linear gain unit 12, which is linearly proportional to the original input signal.

While each of devices 13 and 14 will have some effect over the entire frequency range of interest, it will be readily apparent that the contribution of the integrator may well become negligible at higher frequencies, while that of the differentiator may well become negligible at lower frequencies. It can therefore be considered that the integrator may need only be effective in the region of lower frequencies, for example, below Hz and the differentiator may need only be effective in the region of higher frequencies, such as above 1, Hz.

The improvement provided by the particular circuit illustrated in FIG. The tests which resulted in these performance curves were performed by personnel of the United States National Bureau of Standards in an anechoic chamber located in the Sound Building No. All of the test equipment used was provided by the National Bureau of Standards and was calibrated to achieve a final total error of not more than one db.

For both test runs the speaker sound output was picked up by a microphone placed 2 meters in front of the speaker. The only difference between the test runs resulting in the curves of FIGS. The level of the voltage supplied to the speaker for each test was selected to provide a power level far below the rated maximum for the speaker and to correspond to what would be considered a comfortable listening level. Comparison of the curves shown on FIGS.

Since the difference between speakers depends to a substantial degree on their relative responses at the low frequency and the high frequency ends of the audible frequency range, the comparative tests indicate that use of a device according to the invention can substantially improve the sound reproduction quality provided by a speaker. To confirm the results predicted by the curves of FIG.

Although there is, of course, a certain element of subjectivity in such listening tests, they, at the same time, constitute the ultimate critical measure of the quality of any sound reproduction system. It will further be noted from the curves of FIGS. For many speakers, substantially greater distortion reduction could be achieved by causing the outputs of the differentiator and integrator units to be proportional in magnitude to some power of the time derivative and time integral, respectively, of the original input signal.

More specifically, it would appear that improved results can be achieved if the magnitude of the outputs of these units are proportional to between the first and fourth powers of the differential or integral, respectively. In all cases, the desired algebraic sign, or polarity, at the output of each of the differentiator and integrator units must be the same as the polarity of the derivative of the original signal or integral of the original signal, respectively.

For example if, at a given instant, the derivative of the input signal has a negative polarity, the signal components provided by the differentiator unit to the output of the device must have negative polarity, assuming that no polarity reversal occurs in the linear gain unit providing the component proportional to the original signal.

More likely you might consider the passive integrator has a lower limit frequency perhaps defined by -6dB or more depending on your error tolerance. The true integrator needs an initial condition or a conditional reset or slow leakage to some desired level. The link above shown below gives you some interactive tools to see the difference. Picture from this answer. With different input resistor values there are different unity-gain frequencies for each input.

Picture from this slide-player. A real integrator circuit using real opamps is in fact a first order lowpass with a very low 3dB-cut-off frequency wo caused by the finite open-loop gain of the opamp. However, as far as the integrator function is concerned, this frequency wo could be seen as a kind of "start frequency" for the begin of the integrating property.

But note that real integration means "90 deg phase shift". But in practice there is a broader frequency region which is used for integrating purposes with a phase shift between app This allowable deviation from the ideal phase shift could be used for defining a corresponding frequency range and two "cut-off frequencies" which describe the practical range of integration.

But these "cut-off" frequencies have, of course, nothing to do with any 3dB-values. Sign up to join this community. The best answers are voted up and rise to the top. Stack Overflow for Teams — Start collaborating and sharing organizational knowledge. Create a free Team Why Teams? Learn more. Summing Integrator cut off Ask Question. Asked 1 year, 11 months ago. Modified 1 month ago. Viewed times. Then how do I calculate unity-gain frequency this circuit? Sundark12 Sundark12 55 4 4 bronze badges.

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The selection criteria of the coefficients consist of two points: first the value of coefficient should be within defined minimum and maximum bound, and the absolute value of group delay of the corresponding integrator should be less than 5 samples. In next iteration, another set of coefficients are selected and PARE values are calculated.

This process is repeated till suitable coefficients are obtained. After last iteration, maximum PARE of each iteration is compared and coefficients corresponding to minimum among maximum PARE are the desired coefficients. In the last, all eleven sets of coefficients are listed in a table along with mean and maximum PARE of the corresponding integrator.

By comparison, the optimum coefficients are the one having minimum among maximum PAREs. By substituting these coefficients in 2 , a second-order digital integrator is obtained having magnitude response close to the ideal one and almost linear phase response. The same algorithm is used to design third- and fourth-order digital integrators.

In order to improve the optimization result, PZC optimization method is applied on the result of minimax optimization. The location of poles and zeros of the transfer function is very important for digital system analyses and synthesis. According to the location of poles, it is possible to test stability of the system. Variation of poles and zeros has a significant effect on the response of a design.

The available literature [ 16 , 22 , 23 ] shows that pole-zero optimization is popular among researchers to improve their responses in various fields. The basic general strategy of designing frequency selective filters is also based on pole-zero placements. As placing a pole near the frequency on the unit circle will increase the gain of the frequency response near , similarly, a zero near will diminish the gain near.

Here, as all the three parameters, namely, poles, zeros, and constant, are optimized, the results show tremendous improvement as compared to the previously proposed operators, where only poles and zeros have been optimized. To obtain the poles, zeros, and constant of an integrator , the numerator and denominator polynomials of 1 are rewritten as or The numerator and denominator can be factorized and expressed as where is a scaling constant, are the zeros, and are the poles of integrator , respectively.

For PZC optimization, the modified transfer function can be written as The amplitude of the modified digital integrator is PARE of integrator is defined as Phase response of the modified digital integrator is defined as. Equations 12 — 14 show that the optimization parameters , , and affect not only magnitude but phase response also.

The group delay of the modified integrator is defined as. In this way PZC optimized second-, third-, and fourth-order wideband digital integrators are obtained. Coefficients and maximum PARE for each frequency range for second-, third-, and fourth-order digital integrators are shown in Tables 1 , 2 , and 3 , respectively.

By observing Tables 1 , 2 , and 3 , the minimum of maximum PARE are obtained for the optimum coefficients as mentioned below. Second order: , , , , and. Third order: , , , , , , and. By substituting the above mentioned optimum coefficients in 2 , the resulting transfer functions of second-, third-, and fourth-order digital integrators , , and are obtained as. For comparison, various existing integrators have been considered.

These are Ngo integrator [ 6 ], Gupta et al. Their transfer functions are. The PARE response of the designed integrators and above mentioned integrators over Nyquist frequency range is shown in Figure 3. The useful frequency range, mean, and maximum PAREs in that frequency range for above mentioned integrators are shown in Table 4. The group delay response of the above mentioned integrators is shown in Figure 4. It is seen that the maximum deviations from constant group delay in case of , , and are 0.

It is verified from Figures 3 and 4 and Table 4 that the minimax optimized designed second-, third-, and fourth-order integrators outperform all the existing integrators over entire Nyquist frequency range. The frequency response of the minimax optimized digital integrators is then improved by using pole, zero, and constant PZC optimization.

For this minimax optimized integrators , , and 16 are defined in terms of pole, zero and constant. By substituting all these values in 19 , respectively, the new integrators are obtained as. All the designed integrators 18 and 20 are stable as their respective poles are inside the unit circle.

The group delay response of the designed integrators , , , , , and is shown in Figure 6. It is verified from Figures 5 and 6 and Table 4 that PZC optimization improves frequency both magnitude and phase response of minimax optimized integrators. In this way, a perfect family of second-, third-, and fourth-order wideband stable digital integrators are designed with superior frequency response. The square wave response of designed and existing integrators is shown in Figures 7 and 8 , respectively.

It is seen that the response of designed integrators , , , , , and is exact triangular wave, while the response of existing integrators is distorted triangular wave. A digital differentiator can also be designed by using transfer function of digital integrator in a similar way to that used in the design of analog differentiator, as suggested by Al-Alaoui [ 24 ]. This method consists of four design steps. In the first step, an integrator is designed that has the same range and accuracy as the desired differentiator.

Then, the transfer function of this integrator is inverted and stabilized by reflecting the poles that lie outside the unit circle to inside the unit circle, and in the last step, the magnitude is compensated appropriately. In this section, a family of second-, third-, and fourth-order digital differentiators have been obtained by inverting the transfer functions of designed integrators , , , , , and defined in 18 and 20 , respectively. The resultant transfer functions are.

As all the poles of designed differentiators see 21 are inside the unit circle, therefore stabilization and compensation of these transfer functions are not required. To compare the efficiency of these designed differentiators, various recently proposed differentiators have been considered. These are Ngo differentiator [ 6 ], Gupta et al. The group delay response of the designed differentiators , , , and above mentioned existing differentiators is shown in Figure It is seen that the maximum deviation from constant group delay in case of , , and are 0.

While, the maximum deviation from constant group delay in case of the existing differentiators Ngo [ 6 ], Gupta et al. It is verified from Figures 9 and 10 and Table 5 that the minimax optimized designed second-, third-, and fourth-order differentiators outperform all the existing differentiators over entire Nyquist frequency range. The group delay response of the designed differentiators , , , , , and is shown in Figure It is verified from Figures 11 and 12 and Table 5 that PZC optimization improves frequency both magnitude and phase response of minimax optimized differentiators.

In this way, a perfect family of second-, third-, and fourth-order wideband stable digital differentiators are designed with superior frequency response. The triangular wave response of designed and existing differentiators is shown in Figures 13 and 14 , respectively. It is seen that the response of designed differentiators , , , , , and is exact square wave, while the response of existing differentiators is distorted square wave.

This paper is focused on the use of two optimization methods, namely, minimax and pole, zero, and constant PZC. It has been proved that the efficiency of minimax optimized integrators is remarkably improved by PZC optimization. Subsequently, by modifying the transfer function of these designed integrators appropriately, new differentiators are obtained which have the same accuracy as the designed integrators.

The family of designed recursive integrators and differentiators are important when excellent magnitude and linear phase response is required at the same time. Madhu Jain , 1 Maneesha Gupta, 2 and N. Academic Editor: C. Received 17 Apr Accepted 07 May Published 06 Jun Abstract Proposed work deals with the design of a family of stable IIR digital integrators via use of minimax and pole, zero, and constant optimization methods.

Introduction Digital integrators and differentiators are integral parts of many systems like digital signal processing, control, audio, and video processing, communication, and medical applications. Problem Formulation It is known that IIR integrators have much better magnitude response than FIR integrators of the same order but their phase characteristics are not linear which can cause problems in some of the signal processing applications. The frequency response of the integrator can be obtained from its transfer function by simply evaluating it on the unit circle; that is, Here, is the amplitude and is the phase of the digital integrator, respectively.

The group delay of integrator is defined as The main limitation in the design of IIR digital integrators is to meet the specified magnitude and phase characteristics. Solution Methodologies In this paper, a family of second-, third-, and fourth-order stable wideband digital integrators are designed via use of minimax and pole, zero, and constant optimization. Figure 1. Flowchart of the complete design method for digital integrator. Figure 2. Flowchart of the applied minimax optimization in design of second-order digital integrator.

Table 1. Coefficients and maximum PARE for different frequency ranges of second-order integrator. Table 2. Coefficients and maximum PARE for different frequency ranges of third-order integrator. Table 3. Coefficients and maximum PARE for different frequency ranges of fourth-order integrator.

Table 4. Figure 3. Percentage absolute relative error response of designed integrators; , , and , Ngo integrator [ 6 ], Gupta et al. Figure 4. Group delay response of designed integrators; , , and , Ngo integrator [ 6 ], Gupta et al. Figure 5. Percentage absolute relative error response of designed integrators; , , , , , and. Figure 6. For analysis and design in frequency domain such as the so-called classical method, loopshaping, or Quantitative Feedback Theory QFT , some form frequency response data is needed.

Hence, in this module we show how to formulate a transfer function in Scilab and plot its frequency response. To be concrete, we consider in Figure 1 a simple diagram of robot joint driven by DC motor through a gear transmission with ratio r:1 [1]. Figure 1 robot joint connected to DC motor via a gear transmission. We want to describe a model in transfer function form so that a block diagram can be drawn. To develop the electrical side of DC motor, consider the model shown in Figure 2.

Figure 2 a model of permanent magnet DC motor. From now on we omit the a subscript in the armature inductance and resistance. It is left to the reader to verify that, in Laplace domain, the joint dynamics in Figure 1 can be described by. This can be drawn as a block diagram in Figure 3. Figure 3 block diagram of the robot joint dynamics in Figure 1. So the transfer functions in 5 and 6 reduce to. These two equations correspond to second order differential equation in time domain.

The reduced block diagram of 10 can be drawn as in Figure 4. Figure 4 reduced block diagram of robot joint dynamics. So, the transfer function for a robot joint driven by DC motor we will be using in our study modules is in the form.

Hence the resulting transfer function becomes. Now we demonstrate how to construct a transfer function such as 12 in Scilab. One method begins by creating the Laplace variable s. So far so good. It already looks like However, this data format still lacks some inside information necessary for further processing such as frequency response plot. So one more step is needed to convert it to a continuous-time linear transfer function, by using the syslin command.

The interactive response shown in Scilab console does not look any different than before. Now P is ready to be used by other commands such as bode, freqresp etc. Note that the row vector passed to poly , representing the polynomial coefficients, starts with the lowest order of s.

Xcos is a simulation engine that comes with Scilab. Here we assume that the reader has some familiarity with basic functionality of Xcos to the level that she could create some simple diagram using blocks from standard palettes. The standard block for transfer function can be located in Continuous time systems palette, under the name CLR. When a user clicks on this block, a parameter dialog window emerges. For this block, numerator and denominator in terms of polynomial of s can be put into the input field directly.

For our transfer function 12 , the data is typed in as shown in Figure 5. Figure 5 A dialog window for setting transfer function parameters. In the next section, we will connect some input to this plant and measure its output by an oscilloscope. Figure 6 depicts frequency response concept in a nutshell. This pair of data through out a range of frequency, actually a vector of complex numbers, constitutes a frequency response for an LTI system. Figure 6 An LTI system driven by sinusoid input.

Note that the transfer function used is the DC motor model from the voltage input to shaft velocity output; i. Figure 7 sinusoid input and output comparison. See the input and output comparison from the scope. Change the input frequency by clicking on and put in new value. Do you see the change in amplitude and phase of the output? Now we can plot the magnitude and phase versus frequency by using the following set of commands. Vectors for keeping magnitude and phase information are initialized Figure 8 a simple frequency response plot.

Then, the frequency response of P is computed at each frequency point , converted to polar form, and put into the magnitude and phase vectors. The phase vector is converted from radian to degree. Last, the magnitude and phase are plotted in the same figure using subplot command. Some drawback of this plot is obvious.

The frequency range is quite limited and the plots are concentrated only on narrow low frequency region. Slightly adjust the above Scilab code to yield a much nicer plot range as in Figure 9. Figure 9 a frequency response plot that covers broader frequency range. This type of frequency response is known as Bode plot. An easier way to plot from a transfer function created by syslin is by the command bode. This yields the plot in Figure Type help bode to see options such as how to adjust frequency range.

Figure 10 frequency response using Scilab bode command. Keyword: the frequency range that a plant is responsive is called bandwidth. Strictly speaking, bandwidth covers frequency region such that the gain is above 0. When trying to identify bandwidth from a Bode plot, we can roughly indicate the frequency point where the magnitude curve touches 0 dB line. For example, from Figure 10, the bandwidth is about 0. Another type of frequency response useful for control design is called a Nyquist plot.

Actually, it is the same data expressed in different format. Recall that each point of frequency response is just a complex number. When it is described in polar form, we get a Bode plot.

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