CONTENTS
1.INTRODUCTION
2.PRACTICAL OPERATION
3.DAC TYPES
4.CKT. DIAGRAM
5.APPLICATIONS
6.ADVANTAGES & DISADVANTAGES
7.CONCLUSION
8.BIBLIOGRAPHY
INTRODUCTION
A Digital to analog converter is a device for converting digital signal code to an analog signal(current,voltage or electric charge).A dac inputs a binary number and output an analog voltage or current signal. A DAC converts an abstract finite-precision number (usually a fixed-point binary number) into a concrete physical quantity (e.g., a voltage or a pressure). In particular, DACs are often used to convert finite-precision time series data to a continually-varying physical signal.
A typical DAC converts the abstract numbers into a concrete sequence of impulses that are then processed by a reconstruction filter using some form of interpolation to fill in data between the impulses. Other DAC methods (e.g., methods based on Delta-sigma modulation) produce a pulse-density modulated signal that can then be filtered in a similar way to produce a smoothly-varying signal.By the Nyquist–Shannon sampling theorem, sampled data can be reconstructed perfectly provided that its bandwidth meets certain requirements (e.g., a baseband signal with bandwidth less than the Nyquist frequency). However, even with an ideal reconstruction filter, digital sampling introduces quantization error that makes perfect reconstruction practically impossible. Increasing the digital resolution (i.e., increasing the number of bits used in each sample) or introducing sampling dither can reduce this error.
A DAC converts an abstract finite-precision number (usually a fixed-point binary number) into a concrete physical quantity (e.g., a voltage or a pressure). In particular, DACs are often used to convert finite-precision time series data to a continually-varying physical signal.
A typical DAC converts the abstract numbers into a concrete sequence of impulses that are then processed by a reconstruction filter using some form of interpolation to fill in data between the impulses. Other DAC methods (e.g., methods based on Delta-sigma modulation) produce a pulse-density modulated signal that can then be filtered in a similar way to produce a smoothly-varying signal.By the Nyquist–Shannon sampling theorem, sampled data can be reconstructed perfectly provided that its bandwidth meets certain requirements (e.g., a baseband signal with bandwidth less than the Nyquist frequency). However, even with an ideal reconstruction filter, digital sampling introduces quantization error that makes perfect reconstruction practically impossible. Increasing the digital resolution (i.e., increasing the number of bits used in each sample) or introducing sampling dither can reduce this error.
Connecting digital circuitry to sensor devices is simple if the sensor devices are inherently digital themselves. Switches, relays, and encoders are easily interfaced with gate circuits due to the on/off nature of their signals. However, when analog devices are involved, interfacing becomes much more complex. What is needed is a way to electronically translate analog signals into digital (binary) quantities, and vice versa. An analog-to-digital converter, or ADC, performs the former task while a digital-to-analog converter, or DAC, performs the latter.
An ADC inputs an analog electrical signal such as voltage or current and outputs a binary number. In block diagram form, it can be represented as such: A DAC, on the other hand, inputs a binary number and outputs an analog voltage or current signal. In block diagram form, it looks like this: Together, they are often used in digital systems to provide complete interface with analog sensors and output devices for control systems such as those used in automotive engine controls:
It is much easier to convert a digital signal into an analog signal than it is to do the reverse. Therefore, we will begin with DAC circuitry and then move to ADC circuitry.
PRACTICAL OPERATION
Instead of impulses, usually the sequence of numbers update the analogue voltage at uniform sampling intervals.
These numbers are written to the DAC, typically with a clock signal that causes each number to be latched in sequence, at which time the DAC output voltage changes rapidly from the previous value to the value represented by the currently latched number. The effect of this is that the output voltage is held in time at the current value until the next input number is latched resulting in a piecewise constant or 'staircase' shaped output. This is equivalent to a zero-order hold operation and has an effect on the frequency response of the reconstructed signal.
Piecewise constant signal typical of a zero-order (non-interpolating) DAC output.
The fact that practical DACs output a sequence of piecewise constant values or rectangular pulses would cause multiple harmonics above the nyquist frequency. These are typically removed with a low pass filter acting as a reconstruction filter.
However, this filter means that there is an inherent effect of the zero-order hold on the effective frequency response of the DAC resulting in a mild roll-off of gain at the higher frequencies (often a 3.9224 dB loss at the Nyquist frequency) and depending on the filter, phase distortion. Not all DACs have a zero order response however. This high-frequency roll-off is the output characteristic of the DAC, and is not an inherent property of the sampled data.\
DAC CKT. DIAGRAM
THERE ARE TWO MAIN STAGES
This part have been explained in detail in the previous section, its purpose is to create the voltage V1 which is equivalent to the weight of the binary number on the lines (D0 to D7). Now that this is a resistor network, if we apply any load on the output of the first stage, this load will be considered as an additional resistor in the network, and thus will disturb the network which will no longer provide the correct & desired output voltage. Therefore, to overcome this problem, we need a voltage buffer, here is where the next stage comes...
This stage will isolate the point V1 from the final output V2, while always keeping the voltage V2 at the exact same value of V1. This is what we call a voltage buffer. for the voltage buffer we use an opamp with the output connected to the inverting input (this special configuration of the Op Amp is also called Voltage Follower). The most important things to note are:
1
2
3
DAC types
The most common types of electronic DACs are:
1.
pulse width modulator, the simplest DAC type. A stable current or voltage is switched into a low pass analog filter with a duration determined by the digital input code. This technique is often used for electric motor speed control, and is now becoming common in high-fidelity audio.
2.
Oversampling DACs or interpolating DACs such as the delta-sigma DAC, use a pulse density conversion technique. The oversampling technique allows for the use of a lower resolution DAC internally. A simple 1-bit DAC is often chosen because the oversampled result is inherently linear. The DAC is driven with a pulse density modulated signal, created with the use of a low-pass filter, step non-linearity (the actual 1-bit DAC), and negative feedback loop, in a technique called delta-sigma modulation. This results in an effective high-pass filter acting on the quantization (signal processing) noise, thus steering this noise out of the low frequencies of interest into the high frequencies of little interest, which is called noise shaping (very high frequencies because of the oversampling). The quantization noise at these high frequencies are removed or greatly attenuated by use of an analog low-pass filter at the output (sometimes a simple RC low-pass circuit is sufficient). Most very high resolution DACs (greater than 16 bits) are of this type due to its high linearity and low cost. Higher oversampling rates can either relax the specifications of the output low-pass filter and enable further suppression of quantization noise. Speeds of greater than 100 thousand samples per second (for example, 192 kHz) and resolutions of 24 bits are attainable with Delta-Sigma DACs. A short comparison with pulse width modulation shows that a 1-bit DAC with a simple first-order integrator would have to run at 3 THz (which is physically unrealizable) to achieve 24 meaningful bits of resolution, requiring a higher order low-pass filter in the noise-shaping loop. A single integrator is a low pass filter with a frequency response inversely proportional to frequency and using one such integrator in the noise-shaping loop is a first order delta-sigma modulator. Multiple higher order topologies (such as MASH) are used to achieve higher degrees of noise-shaping with a stable topology.
3.
binary weighted DAC, which contains one resistor or current source for each bit of the DAC connected to a summing point. These precise voltages or currents sum to the correct output value. This is one of the fastest conversion methods but suffers from poor accuracy because of the high precision required for each individual voltage or current. Such high-precision resistors and current-sources are expensive, so this type of converter is usually limited to 8-bit resolution or less.
4.
R-2R ladder DAC, which is a binary weighted DAC that uses a repeating cascaded structure of resistor values R and 2R. This improves the precision due to the relative ease of producing equal valued matched resistors (or current sources). However, wide converters perform slowly due to increasingly large RC-constants for each added R-2R link.
5.
thermometer coded DAC, which contains an equal resistor or current source segment for each possible value of DAC output. An 8-bit thermometer DAC would have 255 segments, and a 16-bit thermometer DAC would have 65,535 segments. This is perhaps the fastest and highest precision DAC architecture but at the expense of high cost. Conversion speeds of >1 billion samples per second have been reached with this type of DAC.
6.
Hybrid DACs, which use a combination of the above techniques in a single converter. Most DAC integrated circuits are of this type due to the difficulty of getting low cost, high speed and high precision in one device.
7.
segmented DAC, which combines the thermometer coded principle for the most significant bits and the binary weighted principle for the least significant bits. In this way, a compromise is obtained between precision (by the use of the thermometer coded principle) and number of resistors or current sources (by the use of the binary weighted principle). The full binary weighted design means 0% segmentation, the full thermometer coded design means 100% segmentation.
DAC performance
DACs are at the beginning of the analog signal chain, which makes them very important to system performance. The most important characteristics of these devices are:
Resolution: This is the number of possible output levels the DAC is designed to reproduce. This is usually stated as the number of bits it uses, which is the base two logarithm of the number of levels. For instance a 1 bit DAC is designed to reproduce 2 (21) levels while an 8 bit DAC is designed for 256 (28) levels. Resolution is related to the effective number of bits (ENOB) which is a measurement of the actual resolution attained by the DAC.
- Maximum sampling frequency: This is a measurement of the maximum speed at which the DACs circuitry can operate and still produce the correct output. As stated in the Nyquist–Shannon sampling theorem, a signal must be sampled at over twice the frequency of the desired signal. For instance, to reproduce signals in all the audible spectrum, which includes frequencies of up to 20 kHz, it is necessary to use DACs that operate at over 40 kHz. The CD standard samples audio at 44.1 kHz, thus DACs of this frequency are often used. A common frequency in cheap computer sound cards is 48 kHz—many work at only this frequency, offering the use of other sample rates only through (often poor) internal resampling.
- Monotonicity: This refers to the ability of a DAC's analog output to move only in the direction that the digital input moves (i.e., if the input increases, the output doesn't dip before asserting the correct output.) This characteristic is very important for DACs used as a low frequency signal source or as a digitally programmable trim element.
- THD+N: This is a measurement of the distortion and noise introduced to the signal by the DAC. It is expressed as a percentage of the total power of unwanted harmonic distortion and noise that accompany the desired signal. This is a very important DAC characteristic for dynamic and small signal DAC applications.
- Dynamic range: This is a measurement of the difference between the largest and smallest signals the DAC can reproduce expressed in decibels. This is usually related to DAC resolution and noise floor.
DAC figures of merit
- Static performance:
- Differential non-linearity (DNL) shows how much two adjacent code analog values deviate from the ideal 1LSB step [1]
- Integral non-linearity (INL) shows how much the DAC transfer characteristic deviates from an ideal one. That is, the ideal characteristic is usually a straight line; INL shows how much the actual voltage at a given code value differs from that line, in LSBs (1LSB steps).
- Gain
- Offset
- Noise is ultimately limited by the thermal noise generated by passive components such as resistors. For audio applications and in room temperatures, such noise is usually a little less than 1 μV (microvolt) of white noise. This limits performance to less than 20~21 bits even in 24-bit DACs, and cannot be corrected unless one resorts to extremely low temperatures to create superconductivity: clearly an impractical proposition.
- Frequency domain performance
- Spurious-free dynamic range (SFDR) indicates in dB the ratio between the powers of the converted main signal and the greatest undesired spur
- Signal to noise and distortion ratio (SNDR) indicates in dB the ratio between the powers of the converted main signal and the sum of the noise and the generated harmonic spurs
- i-th harmonic distortion (HDi) indicates the power of the i-th harmonic of the converted main signal
- Total harmonic distortion (THD) is the sum of the powers of all HDi
- If the maximum DNL error is less than 1 LSB, then D/A converter is guaranteed to be monotonic.
Applications
Audio
Top-loading CD player and external digital-to-analog converter.
Most modern audio signals are stored in digital form (for example MP3s and CDs) and in order to be heard through speakers they must be converted into an analog signal. DACs are therefore found in CD players, digital music players, and PC sound cards.
Specialist stand-alone DACs can also be found in high-end hi-fi systems. These normally take the digital output of a CD player (or dedicated transport) and convert the signal into a line-level output that can then be fed into a pre-amplifier stage.Similar digital-to-analog converters can be found in digital speakers such as USB speakers, and in sound cards.
Video
Video signals from a digital source, such as a computer, must be converted to analog form if they are to be displayed on an analog monitor. As of 2007, analog inputs are more commonly used than digital, but this may change as flat panel displays with DVI and/or HDMI connections become more widespread. A video DAC is, however, incorporated in any Digital Video Player with analog outputs. The DAC is usually integrated with some memory (RAM), which contains conversion tables for gamma correction, contrast and brightness, to make a device called a RAMDAC
RESOLUTIONN FEATURE
An important quality feature of a digital/analog converter (DAC) is its resolution. Sigma-delta modulation (delta-sigma modulation) provides a high resolution digital-to-analog conversion solution. Sigma-delta DACs have come into widespread use with the development of signal processing and digital audio technologies and their applications. A delta-sigma DAC has a digital input summer, a digital interpolation filter, a digital feedback loop, a quantizer, and a DAC output stage at the modulator output. Sigma-delta DACs commonly include a front-end interpolator which receives digital input samples and increases the sampling rate of the digital input samples. The sigma-delta modulator receives the higher frequency input samples from the interpolator and converts the samples to a lower resolution (typical one-bit), high frequency bit stream. Rather than spreading quantization noise uniformly over the frequency range from 0 to the sampling Nyquist frequency, the sigma delta modulator shapes the noise so that the majority of the noise falls into the very high frequencies above the Nyquist frequency. The performance of a digital-to-analog converter in audio equipment is principally represented by factors such as a distortion factor (a ratio of a harmonic component to a signal) and a signal to noise (SIN) ratio. Sigma-delta modulators, which can be used in sigma-delta converters, can provide a degree of shaping (filtering) of quantization noise that can be present. The higher the order of the sigma-delta modulator, the further the quantization noise is pushed into the frequency band and the greater the separation between the signal being converted and the quantization noise. Delta-sigma modulators are particularly useful in digital-to-analog converter (DAC) systems. Using oversampling, a delta-sigma modulator spreads the quantization noise power across the oversampling frequency band, which is typically much greater than the input signal bandwidth.
ADVANTAGES OF DAC
The advantage of Digital to analog converters
A digital to analog converter has drawbacks, but it has advantages of allowing signals to be controlled and processed to the best of speed and precision with the use of a micro processor. One can generate a perfect sine or triangular wave with a microprocessor and convert it into a real analog wave. One can process audio with a CPU or DSP and convert it back to audio. Once you have audio inside a processor the sky is the limit to what you can do with it and with custom firmware.
Then the use is also obvious for precision digital instrumentation such as reading a load cell, thermo-couple or any sensor, converting it into an industry standard 4-20mA , 0-20mA or 0-5V or 0-10V output for PLCs or other instruments to interface with.
Digital to analog converters may offer an economical and compact way to have precision signals. For a 24bit resolution, an 16.7million divisions of it's full scale deflection can be expected and perform thousands of conversions per second.
The drawbacks are that they don't always produce perfectly smooth outputs, which may be subjected to some quantisation noise and the use of external filters may be required in some cases.
Very nice technologies are available these days that allow SPI, I2C serial interfacing with high speed, high resolution and compact SMD chipsets at economical prices.
The thing to note is that a digital to analog converter is rather useless without a micro processor to control and feed it with data and commonly they are build into DSP (digital signal processors) already.
DISADVANTAGES OF DAC
PCM conversion circuit 18 enables converting a digital signal having a determined number of bits into a digital signal having a smaller number of bits while rejecting part of the quantization noise introduced by such a conversion outside a usefulfrequency band. The pulse-width modulation being a non-linear process, the conversion of the PCM signal into a PWM signal causes the occurrence of spectral crosstalk components. Since a non-negligible part of the power of the PCM signal provided by PCMconversion unit 18 is present in the form of quantization noise outside of the useful frequency band, the crosstalk components cause a significant increase of the noise level in the useful frequency band on conversion of the PCM signal into a PWM signal. There thus is a degradation of the signal-to-noise ratio at the converter output, in other words, for an audio application, a degradation of the quality of the audio signal provided by loudspeaker 12.
There are two solutions to avoid such disadvantages:
A first solution is to decrease the quantization noise at the output of PCM conversion unit 18 by increasing the number of states that the PCM signal can code. The quantization noise decrease enables decreasing the crosstalk components resultingfrom the PWM modulation, which limits the degradation of the quality of the PWM signal. However, an increase in the number of states that the PCM signal can code implies an increase in the resolution of the PWM signal and thus requires an increase inthe frequency of control signal CLK of PWM conversion unit 20. A disadvantage of such a solution is that the provision of control signal CLK at a high frequency requires use of expensive components, for example, phase-locked loops.
2. Second solution which provides, at the level of feedback loop 21 which provides the PCM signal at the input of PCM conversion unit 18, a unit with a variable transfer function 22 which models the frequency response of PWMconversion unit 20. PCM conversion unit 18, which behaves as a bandpass filter towards the noise introduced into feedback loop 21, thus filters the noise present in the useful frequency band of the signal provided by variable transfer function unit 22,which provides a PWM signal having an acceptable noise level in the useful frequency band.
Relationship between digital and analog data
The Relationship between Analog and Digital Data
The relationship between analog and digital data is best illustrated pictorially. Suppose that we have a 4 bit system, meaning that we have 4 separate digital values by through which we can specify a number. There are 4216= different on/off combinations that could be made through these values. The range of these values is up to us. In this lab we will take the lowest possible analog value that -5V while the largest possible value will be +5V. These will correspond to the bottom of and the top , respectively. In other words, in Fig. 1 is 5V. FS stands for Full Scale. Notice that this outlines 16 different voltage bands marked off in red. Each band corresponds to a different combination of on/off values. These are shown in blue to the left where blue indicated on while white indicates off. The value of a waveform can therefore be indicated with these digital outputs.
Conclusion
Many embedded applications require the generation of analog signals. Although separate digital to
analog converter ICs exist on the market today, Fusion with a PWM, such as CorePWM, can integrate these
components and reduce cost and circuit board space while improving reliability in embedded control
applications that need a PWM. A low cost, simple DAC can be implemented with Fusion, ProASIC®3/E,
ProASICPLUS®, Axcelerator®, or RTAX-S FPGAs using Actel Core8051 and CorePWM plus a few external
components.
These techniques can be used to generate dual-tone multiplexed frequencies (DTMF) for telephone
dialing, controlling the speed of a motor, generating sound and complex waveforms, generating variable
voltages, and performing voltage trimming in a power management system.
BIBLIOGRAPHY
1. http://www.electronics-manufacturers.com/info/data-acquisition/digital-to-analog-converter.html
2. http://wiki.answers.com/Q/What_is_the_advantages_of_digital_analog_converter
3. http://www.patentstorm.us/patents/7283078/description.html
4. http://www.actel.com/ipdocs/CorePWM_DS.pdf
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