News and analytical portal "electronics time". Do you need a separate DAC? What to use as a DAC

An obvious trend in modern household audio equipment is various portable speakers and headphones; it is in these product categories that the largest number of items are represented today. It is very difficult to compete with them in popularity, but there is one device, the need for which is constantly increasing - a DAC, a digital-to-analog converter. Why is it needed?

Let's use the "by contradiction" method. If you are an orthodox conservative and don’t listen to anything other than FM radio, records and other magnetic albums, then you DO NOT need a DAC. For everyone else, from gamers to movie buffs, this is definitely a must have, unless of course you are used to being content with your favorite hobby on a residual basis.

By the way, why is music recorded, stored and transmitted digitally at all? After all, by its nature it is analog. First of all, it’s convenient, since you can’t really carry a record or reel under your armpit. Then, the digital format implies lossless transmission and copying. So the main task of the DAC is to produce the conversion as efficiently as possible.

The simplest example is a typical smartphone. Most of us have a lot of songs stored in it, among other things, or have the ability to stream from the Internet. It would seem that all you have to do is plug in your headphones and enjoy the music. But the standard DAC of a smartphone is not only most often developed by non-audiophiles, but also, as the main point of the technical specifications, it has low power consumption, which does not correlate with sound quality at all. The solution is to use an external converter, portable and long-lasting (due to its own battery), which will be able to “pump up” even the tightest headphones.

But what about at home, where the problem of saving energy is, frankly speaking, secondary? Let's say you like some TV channel or program, play on a console or watch a movie. The audio system of the vast majority of modern flat-screen TVs is developed according to the residual principle, up to the category of “performance monitoring”, much like with standard cables or headphones - make sure that the device is functioning and put them aside. The situation is the same with analog outputs - they are there, but frankly speaking - “for show”. Digital outputs, if they differ in quality, are within much smaller limits. Thus, it is possible to fully connect the TV to an existing stereo system, and this is again the task of the DAC.

For people whose work takes place directly at the computer, the DAC is also a serious help and even joy. By connecting speakers or headphones through it, you can provide yourself with high-quality music in parallel with your work process. There are a lot of similar examples of use, so the question “should/shouldn’t” does not arise here, the task is solely to select a suitable device.

So, whatever one may say, today you simply cannot do without a good DAC.

Digital to analog converters (DACs) — designed to convert digital signals to analog. Such a conversion is necessary, for example, when restoring an analog signal that has been previously converted to digital for long-distance transmission or storage (such a signal, in particular, can be sound). Another example of the use of such a conversion is obtaining a control signal when digitally controlling devices whose operating mode is determined directly by an analog signal (which, in particular, occurs when controlling motors).

(xtypo_quote)The main parameters of the DAC include resolution, settling time, nonlinearity error, etc.(/xtypo_quote)

Resolution is the reciprocal of the maximum number of quantization steps of the output analog signal. The establishment time t set is the time interval from the application of the code to the input until the moment when the output signal enters the specified limits determined by the error. Nonlinearity error is the maximum deviation of the graph of the dependence of the output voltage on the voltage specified by the digital signal in relation to the ideal straight line over the entire conversion range.

Like the ones under consideration, DACs are a “link” between analog and digital electronics. There are various principles for constructing an ADC.

DAC circuit with summation of weight currents

In Fig. Figure 3.88 shows a DAC circuit with summation of weight currents.

Key S 5 is closed only when all keys S 1 ... S 4 are open (in this case u out = 0). U 0

— reference voltage. Each resistor in the input circuit corresponds to a specific bit of a binary number.

Essentially this DAC is an inverting amplifier based on an operational amplifier. Analysis of such a scheme is not difficult. So, if one key is closed

S1, then uout = −U 0 R oc / R

which corresponds to the first and zeros in the remaining digits.

From the analysis of the circuit it follows that the modulus of the output voltage is proportional to the number, the binary code of which is determined by the state of the keys S 1 ... S 4. The currents of the keys S 1 ... S 4 are summed up at point “a”, and the currents of different keys are different (have different “weights”). This determines the name of the scheme.

From the above it follows that u out = − (U 0 R oc / R) S 1 − (U 0 R oc / (R/2)) S 2 - − (U 0 R oc / (R/4)) · S 3 − (U 0 R oc / (R/8)) · S 4 = = − (U 0 R oc / R) · (8S 4 + 4S 3 + 2S 2 + S 1)

where S i ,i = 1, 2, 3, 4 takes the value 1 if the corresponding key is closed, and 0 if the key is open.

The state of the keys is determined by the input converted code. The circuit is simple, but has disadvantages: significant changes in voltage across the switches and the use of resistors with very different resistances. It is difficult to ensure the required accuracy of these resistances.

DAC based on resistive matrix R - 2R

Let's consider a DAC based on a resistive matrix R - 2R (constant resistance matrix) (Fig. 3.89).

The circuit uses so-called changeover switches S 1 ... S 4 , each of which is connected to a common point in one of the states, so the voltages on the keys are low. Key S 5 is closed only when all keys S 1 ... S 4 are connected to a common point. The input circuit uses resistors with only two different resistance values.

From the analysis of the circuit, you can see that for it, the modulus of the output voltage is proportional to the number, the binary code of which is determined by the state of the keys S 1 ... S 4. The analysis is easy to perform given the following. Let each of the keys S 1 ... S 4 be connected to a common point. Then, as is easy to see, the voltage relative to the common point at each subsequent point “a” ... “d” is 2 times greater than at the previous one. For example, the voltage at point “b” is 2 times greater than at point “a” (voltages U a, U b, U c and U d at these points are determined as follows:

Let's assume that the state of the specified keys has changed. Then the voltages at points “a” ... “d” will not change, since the voltage between the inputs of the operational amplifier is practically zero.

From the above it follows that:

u out = − (U 0 R oc / 2R) S 4 − ((U 0 /2) R oc / 2R) S 3 - ((U 0 /4) R oc / 2R) S 2 − (( U 0 /8) R oc / 2R) S 1 = − (U 0 R oc / 16R) (8S 4 + 4S 3 + 2S 2 + S 1)

where S i , i = 1, 2, 3, 4 takes the value 1 if the corresponding key is closed, and 0 if the key is open.

DAC for BCD conversion

Let's consider a DAC for converting binary-decimal numbers (Fig. 3.90).

A separate R − 2R matrix (indicated by rectangles) is used to represent each decimal place. Z 0 …Z 3 denote the numbers determined by the state of the keys of each matrix R − 2R. The principle of operation becomes clear if we consider that the resistance of each matrix is ​​R, and if we analyze the fragment of the circuit presented in Fig. 3.91. From the analysis it follows that

Schemes for the use of digital-to-analog converters relate not only to the field of code-to-analog conversion. Using their properties, you can determine the products of two or more signals, build function dividers, analog links controlled by microcontrollers, such as attenuators, integrators. Signal generators, including arbitrary waveforms, are also an important area of ​​application for DACs. Below are some signal processing circuits that include D-A converters.

Handling signed numbers

Until now, when describing digital-to-analog converters, input digital information was represented in the form of natural numbers (unipolar). Processing integers (bipolar) has certain features. Typically, binary integers are represented using two's complement code. In this way, using eight digits, you can represent numbers in the range from -128 to +127. When entering numbers into the DAC, this range of numbers is shifted to 0...255 by adding 128. Numbers greater than 128 are considered positive, and numbers less than 128 are considered negative. The average number 128 corresponds to zero. This representation of signed numbers is called a shifted code. Adding a number that is half the full scale of a given bit (in our example it is 128) can be easily done by inverting the most significant (sign) bit. The correspondence of the considered codes is illustrated in Table. 1.

Table 1

01111111

00000001

00000000

11111111

10000001

10000000

11111111

10000001

10000000

01111111

00000001

00000000

127/255

1/255

0

-1/255

-127/255

-128/255

To obtain an output signal with the correct sign, it is necessary to reverse shift by subtracting the current or voltage that is half the scale of the converter. This can be done in different ways for different types of DACs. For example, with DACs based on current sources, the range of variation of the reference voltage is limited, and the output voltage has a polarity opposite to the polarity of the reference voltage. In this case, the bipolar mode is most simply implemented by including an additional bias resistor R cm between the DAC output and the reference voltage input (Fig. 18a). Resistor R cm is manufactured on an IC chip. Its resistance is chosen such that the current I cm is half the maximum value of the DAC output current.

In principle, the problem of output current bias can be solved similarly for DACs based on MOS switches. To do this, you need to invert the reference voltage, and then generate a bias current from -Uop, which should be subtracted from the DAC output current. However, to maintain temperature stability, it is better to ensure that the bias current is generated directly in the DAC. To do this, in the diagram in Fig. 8a, a second operational amplifier is introduced and the second output of the DAC is connected to the input of this op-amp (Fig. 18b).

The second output current of the DAC, according to (10),

or, taking into account (8)

(23)

(24)

(25)

In the case of N=8, this coincides with the data in Table 2 up to a factor of 2. 6, taking into account the fact that for a converter based on MOS switches the maximum output current

If resistors R2 are well matched in resistance, then an absolute change in their value with temperature fluctuations does not affect the output voltage of the circuit.

For digital-to-analog converters with an output signal in the form of voltage, built on an inverse resistive matrix (see Fig. 9), the bipolar mode can be more easily implemented (Fig. 18c). Typically, such DACs contain an on-chip output buffer amplifier. To operate the DAC in a unipolar connection, the free terminal of the lower resistor R in the circuit is not connected, or is connected to a common point in the circuit to double the output voltage. To operate in a bipolar connection, the free output of this resistor is connected to the reference voltage input of the DAC. In this case, the op-amp operates in differential connection and its output voltage, taking into account (16)

(26)

Multipliers and dividers of functions

As mentioned above, D-A converters based on MOS switches allow changes in the reference voltage within a wide range, including a change in polarity. From formulas (8) and (17) it follows that the DAC output voltage is proportional to the product of the reference voltage and the input digital code. This circumstance makes it possible to directly use such DACs to multiply an analog signal by a digital code.

When the DAC is connected unipolarly, the output signal is proportional to the product of a bipolar analog signal and a unipolar digital code. Such a multiplier is called a two-quadrant multiplier. When the DAC is connected bipolarly (Fig. 18b and 18c), the output signal is proportional to the product of a bipolar analog signal and a bipolar digital code. This circuit can work as a four-quadrant multiplier.

Dividing the input voltage by a digital scale M D =D/2 N is performed using a two-quadrant divider circuit (Fig. 19).

In the diagram in Fig. 19a, a MOS switch converter with a current output operates as a voltage-to-current converter controlled by code D and included in the feedback circuit of the op-amp. The input voltage is applied to the free terminal of the DAC feedback resistor located on the IC chip. In this circuit, the output current of the DAC is

that when the condition R os = R is fulfilled, it gives

It should be noted that with the code "all zeros" the feedback is opened. This mode can be prevented by either disabling such code in software, or by connecting a resistor with a resistance equal to R·2 N+1 between the output and the inverting input of the op-amp.

A divider circuit based on a DAC with a voltage output built on an inverse resistive matrix and including a buffer op-amp is shown in Fig. 8.19b. The output and input voltages of this circuit are related by the equation

(27)

this implies

In this circuit, the amplifier is covered by both positive and negative feedback. For negative feedback to prevail (otherwise the op-amp will turn into a comparator), condition D must be met<2 N-1 или M D <1/2. Это ограничивает значение входного кода нижней половиной шкалы.

Attenuators and integrators on DACs

Attenuators, i.e. Digitally controlled signal level regulators are much more reliable and durable than traditional attenuators based on variable resistors. It is advisable to use them in measuring instruments and other devices that require adjustment of parameters, especially automatic ones. Such attenuators can most simply be built on the basis of a multiplying DAC with an inverse resistive matrix and a buffer amplifier. In principle, any DAC of the specified type is suitable for this purpose, but some companies produce converters optimized to perform this function. In Fig. Figure 20a shows an attenuator circuit using a variable resistor, and Fig. 20b - a similar circuit on a multiplying DAC.

If the input signal is unipolar, it is advisable to use a single-supply DAC, but the buffer op-amp must have a rail-to-rail output, i.e. its output voltage must reach zero and the supply voltage. If the DAC is multi-channel, then each converter on the chip must have an individual reference voltage input. These requirements are met to varying degrees by such DAC ICs as 2-channel 12-bit MAX532, 4-channel 8-bit MAX509, 8-channel 8-bit AD8441, 8-channel 8-bit DAC-8841, etc. .

To build an integrator with a digital setting of the integration time constant, you can use the basic integrator circuit, and use a DAC with voltage summation as an input resistor (Fig. 12). Based on such a circuit, filters can be built, including filters based on the state variable method, tunable pulse generators, etc.

Direct digital signal synthesis systems

An important area of ​​application for DACs is the synthesis of analog signals of the required shape. Analog signal generators - sinusoidal, triangular and rectangular shapes - have low accuracy and stability, and cannot be controlled by a computer. In recent years, systems for direct digital signal synthesis have been developed, providing high accuracy in setting the frequency and initial phase of signals, as well as high fidelity in reproducing their shape. Moreover, these systems allow generating signals of a wide variety of shapes, including user-defined shapes. A simplified block diagram of a direct digital signal synthesis generator is shown in Fig. 21.

In principle, direct digital synthesis systems are simple. Moreover, the theory and basic methods for constructing such systems have been known for about 30 years. True, only recently have DACs and specialized analog-to-digital ICs appeared that are suitable for synthesizing signals over a wide frequency band.

The direct digital synthesis circuit contains three main blocks: a phase angle generator, memory and DAC. The phase angle generator is typically an accumulator with a register. It works simply as a phase register, the contents of which are incremented by a certain phase angle at specified time intervals. The phase increment Dj is loaded as a digital code into the input registers. Memory plays the role of a function table. The code of the current phase is supplied to its address inputs, and from the data output to the input of the DA converter a code corresponding to the current value of the specified function is received. The DAC, in turn, generates an analog signal.

The register contains the current phase of the output signal as an integer, which, when divided by 2N, where N is the size of the adder, is equal to the fraction of the period. Increasing the register bit depth only increases the resolution of this portion. The frequency of the output signal is equal to the product of the clock frequency f clock and the phase increment in each clock period. When using an N-bit adder, the frequency of the output signal will be equal to

Direct synthesis generators are available in the form of ICs. In particular, the AD9850 chip, the simplified structure of which is shown in Fig. 21, contains a 32-bit phase angle generator and a 10-bit DAC. The phase increment is loaded via the 8-bit data bus, byte-by-byte, into four input registers. The memory contains a table of sines. The maximum allowed clock frequency is 125 MHz. In this case, the frequency resolution is 0.0291 Hz. The fast interface allows you to change the output signal frequency up to 23 million times per second.

Let's look into this topic and find out for ourselves once and for all whether you need a separate DAC or not.

I think you have read more than once on forums or heard from friends that in order to listen to high-quality sound you simply need to buy a separate DAC. Whether this is true or not, we will find out. Do you really need a DAC, how much does it cost and how does it work - I will tell you about all this in this article. A small spoiler - it may turn out that you don’t need a separate DAC at all.

So what is a DAC?

The DAC converts the digital signal to analog so your headphones can produce sound. It's that simple! Most DAC chips are found in the sources of whatever you plug your headphones into and usually cost between $3 and $30 to the manufacturer. This is a very simple and permanent component of any smartphone, although they are trying to kill the headphone jack, mainly by Apple.

Like headphone amplifiers, separate DACs began to appear on sale in order to somehow compensate for the low level of sound reproduction quality. You will be surprised, but in the 80s and even in the 90s, not all consumer equipment could handle even simple headphones, not to mention more serious equipment. Quite often, even if the DAC was built into the finished device, it was not connected to it correctly or was poorly shielded, so while listening you could hear interference or interference from the operation of the equipment itself. Add here the far from best quality of musical material with a low sampling rate of the first mp3 files and you can imagine what the music of the 90s was like. Agree, no one wants to listen to this for their own pleasure.

The DAC converts the digital signal to analog so your headphones can produce sound. It's that simple!

But digital music has come a long way since then. The significantly increased culture of musical equipment production has led to the fact that even cheap chips began to provide fairly good sound quality, and musical material has become much higher quality. Today, music almost everywhere is recorded with a bitrate of 320 kb/sec, and many have switched to listening to material in or in the newfangled . And where previously serious equipment was required to achieve good sound quality, now in most cases the capabilities of an ordinary good smartphone are sufficient.

How does a DAC work?

Any audio, no matter how it is stored, in the form of a vinyl record or mp3 file, is a sound wave. When a computer records an analog signal, it represents it in digital form, resembling an analog wave. But if the analog sound wave is smooth and continuous, then its digital version is discrete. This means that the wave is not recorded continuously, but once in a certain period of time. The Y axis records the amplitude of the wave (i.e. how loud it is), and the X axis records its change over time. Each wave has a certain period that changes over time, and this is called frequency, measured in Hertz. I think you've already heard somewhere that the frequency of a wave determines the tone of the sound you hear. The higher the frequency (i.e., the more often per unit time the wave takes its maximum and minimum values), the higher the tone of the sound.

The DAC's job is to receive digitized sound information and use it to recreate the original sound in analog form, which is then fed into the headphones and you hear the sound. To do this, the DAC receives data about each position of the sound wave many times per second, this value is called the sampling frequency, the higher it is, the closer the digital copy is to the original and the higher the sound quality. Due to the fact that DACs are imperfect, various problems may arise during this procedure of converting a digital signal to analogue, these are: jittering, narrow dynamic range And limited bitrate(low sampling rate).

The DAC's job is to receive digitized sound information and use it to recreate the original sound in analog form, which is then fed into the headphones and you hear the sound.

Before you continue, you should remember the following terms: bitrate, audio recording bit rate And sampling frequency.

Bitrate— shows how much information about sound is recorded for one second of sound.

— shows how many times per second the change in the amplitude of the sound signal was measured.

Sound bit rate— shows how much data was recorded during one sampling rate measurement.

What is Jittering?

But still, what is jitter? This effect is entirely dependent on the sampling rate, or how often we measure the change in amplitude of an analog sound wave. Imagine we do this less frequently than 44.1 Hz or once per second. If we try to digitize the sound of a very high frequency in this way, for example, drum kit cymbals or bells, we will not have time to measure the passage of the full amplitude of the sound by the signal and, if the circumstances are unsuccessful, we will measure only the lower amplitude values ​​or the middle ones. As a result, instead of a high and clear sound, we get an indistinct trembling of sound, which is not at all similar to what we recorded. Just look at the illustration and everything will become clear to you.

The minimum required sampling frequency for complete absence of jitter is 44.1 Hz.

Under the item A you see the normal sampling frequency, which manages to measure the movement of the sound wave in each of its positions and in digitized form we will get the same sound as we heard from a live instrument. On the image B We see that the amplitude of the sound has time to completely change, but the sampling frequency is not sufficient to measure this, and therefore we will hear a jitter of the sound at a lower frequency than the original sound.

If you listen to old mp3 files or bad MIDI recordings, you will notice that it is difficult for you to distinguish between musical instruments, if they are playing simultaneously, they simply merge into a “sound mess” and it is impossible to make out anything in it.

This is because the recording has a narrow dynamic range.. The larger it is, the deeper the sound is heard, more pleasant and realistic. The narrow dynamic range simply does not allow different instruments that sound at the same time to have different volumes and one instrument drowns out the other, which results in a muddy, unpleasant sound and you don’t want to listen to such music at all.

Low bitrate of audio recording is to blame for the narrow dynamic range.

Theoretically, the dynamic range is determined by the bit depth of the sound during its encoding into digital form.. The higher the bit rate, the more values ​​a sound wave can take per unit of time and the wider the dynamic range can be. But this is in theory, because... In addition to the bit rate, the volume can be influenced by many other factors, and the bit rate begins to affect the dynamic range when all other factors are excluded.

For example, almost all modern music is released with significant compression to increase the base volume of all material, the dynamic range suffers greatly because all quiet parts of the composition are tightened up and become louder, and very loud instrument peaks are cut down to a medium value. Thus, after the compression procedure, it almost doesn’t matter what the bit depth of the recording was. But if you are listening to high-quality material that has not been ruined in the studio, the bit rate really begins to play a significant role in the dynamic range.

Remember, the higher the bit rate, the more values ​​of the sound wave volume level are recorded per unit of time, and the wider the dynamic range.

The most common value today is 16-bit recording, but 24-bit music is already gaining popularity, and soon 32-bit recordings of musical works will begin to come into general use. With high-quality processing of musical material in the studio and without terrible compression, 16-bit recording accuracy is generally sufficient to avoid problems with dynamic range.

But in determining sound quality we are again faced with the peculiarities of human perception of sound. What is 16 bit audio recording? This means that one measurement of the change in amplitude of a sound wave can take 65536 values, which gives us a dynamic range of up to 96.33 dB. In turn, this means that sound with a volume of up to 96.33 dB must be recorded without distortion in volume level.

If you're like me, most of the time you listen to music on headphones, and headphones can be quite dangerous to listen to loud music for long periods of time, and believe me, 96.33 dB is very loud. I try not to exceed 60-65 dB when listening, this is enough to fully enjoy the sound, but not enough to damage your hearing. And, as you can see, I still have a significant volume margin up to the coveted 96.33 dB. For this reason, recordings with 24-bit precision will not provide any advantage for me, I simply will not hear the difference due to the fact that I do not listen to the music loud enough. If someone you know who listens to music on headphones tells you that there is a difference between 16-bit recording and 24-bit recording, don’t believe him. He has become a victim of marketing and simply believes that there is a difference, even though he cannot hear it. Let's add to this the fact that our hearing has different sensitivity in volume to different sound frequencies, so 16-bit recordings for listening on headphones are enough for all situations.

16-bit recording allows us to record the signal volume in the form of 65536 values, which gives us a volume level of 96.33 dB.

So why do many people believe that 24-bit music recording is significantly superior to 16-bit? For some situations this is true. For example, if you are listening to a live recording of a symphony orchestra, you really need a 24-bit recording because... you'll have to turn up the volume significantly to hear all the nuances. You're turning up the volume, technically, on your device, but the volume you hear will be normal, because symphonic music recordings are kept quite quiet just so that you can hear all the nuances of the sound. But this rule does not work for modern pop music recordings, because... Already in the studio, recordings are made extremely loud, and if you listen to it at the same volume as a high-quality orchestra recording, you simply risk damaging your hearing.

24-bit recording is also suitable for audio recording. It is much more efficient to record at a higher bit rate and then, during final processing, reduce it to 16, than vice versa. If you record in 16 bits and then artificially increase it to 24, the quality will be even lower than the original 16 bits, and it is possible that extraneous background noise will appear in the sound.

What should the bitrate be?

Many of you must have noticed a difference in the sound quality of songs if you first listened to them on YouTube and then switched to listening to CDs or even iTunes. This difference in quality is due to the bitrate. In general, in everyday life, in 95% of cases, the quality of the recording is determined by the bitrate.

As we remember, bitrate is the amount of information about a sound wave that we record per unit of time, most often per 1 second.

You remember that the higher the bit rate of the sound, the more information we need to record per unit of time about each value of the sound amplitude measurement, the more bitrate we need. That's why bitrate is so important, it doesn't matter the bit rate of the audio, if you have a small bitrate, you simply won't be able to record all the necessary data for high-quality playback.

Just remember - the higher the bitrate, the higher the sound quality. It's that simple.

For most cases, 320 Kb/sec is enough. Moreover, most people simply will not notice the difference if they listen to music with a high bitrate.

I am often asked what the bitrate should be for high-quality sound? I answer - 320 Kb/Sec will be enough, provided that you have a 16-bit recording. Yes, when listening to FLAC files with the same music, you can hear the difference and FLAC will sound better, however, in order to hear this difference you need good headphones and sound equipment, and also that it is quiet around you. Those. you need good home conditions. For home use, I recommend storing music in FLAC format with a higher bitrate and bit rate, but for any mobile use (and most of us listen to music on the go), an mp3 file with a bitrate of 320 Kbps is more than enough. In addition, the memory on your mobile device is limited and listening to mp3 saves it perfectly, or your mobile traffic if you listen to music using streaming services.

So do I need a DAC or not?

Probably not, you don't need a separate DAC. The only reason to buy a DAC today is that your computer, smartphone or home audio system doesn't have one, or your devices use old and very cheap built-in DACs, and the sound quality when connecting any headphones is terrible. Then yes, you need a DAC. But if you are using modern equipment, then you do not need a separate DAC, you simply will not hear the difference.

Probably not, you don't need a separate DAC.

Another reason to buy a separate DAC is that you are building a true high-end sound system, where every element must be flawless. In this case I agree, you need a separate DAC.

In most cases, what is already built into your phone, tablet or laptop will be enough for you, modern DACs are good enough so that you do not hear noticeable sound distortion and for 80% of our audience I can safely say - don’t buy DACs, you just you won't hear the difference.

Vinyl, of course, is a fashionable thing now, friends, but it will never have to overcome digital music distribution. For more than a decade and a half, digital audio sources have firmly held a dominant position in both the professional and consumer electronics sectors. Let's talk about how to squeeze the maximum Hi-Fi juice out of an assortment of fruits - from Internet radio stations to 24-bit audio.

Once upon a time, the CD player was the only solution, and was generally considered cool High End at first, but today this topic seems to be considered morally exhausted. Yes, in the old fashioned way, many still keep CDs in their collections, but as a physical medium it is inferior to vinyl, which simply looks more beautiful, and is technically inferior in terms of parameters to HD audio, which is already widely sold on the Internet not only by audiophiles, but also by major labels. So, instead of a CD player, we need a more versatile device with external inputs that could convert the binary code of zeros and ones into an analog signal that would then be fed to the amplifier and speakers eventually.

DACs are everywhere

An AV receiver, a CD, and, in principle, any media player are equipped with a unit with a digital-to-analog converter (DAC, converter, DAC). As an independent device, DACs appeared as a High-End upgrade to an existing CD player. The designers believed that it would be wiser to separate the player into separate units with their own power supply.

One of the first external DACs Sony DAS-R1, released at the end of 1987

In the first, the actual mechanical part with a readout optical system and a digital output was installed. It was called CD transport. In the second block there were no longer any moving nodes - only a DAC board, the importance of which has now grown to the title of a digital hub. By the way, it often happens that a modern CD player has a pair of digital inputs for connecting external sources.

The life cycle of sound from the source, subsequent recording and digitization, processing, and the reverse cycle - digital-to-analog conversion

A modern converter interacts with a number of signal sources - the main thing is that there is appropriate switching for everyone. The source can also be an old DVD player - they are usually connected via optical TosLink or coaxial cable. The latter looks like an ordinary “tulip” from a stereo pair. Expensive models may also use XLR connectors. Using the USB input, you can connect a computer or portable audio source to the DAC.

In addition, portable DACs are made compatible with sources based on iOS or Android phones, iPods, tablets and other gadgets. In fact, in all these cases, the converter becomes an external sound module with separate power supply and good hardware, which are unheard of in standard multimedia equipment. And modern DACs are often equipped with a headphone amplifier.

Multi-bit and single-bit DACs

Until the 21st century, digital-to-analog converters only handled 16-bit audio, according to the Red Book CD format. There was simply no other way. The sampling frequency for CDs was 44 kHz, while for professional DAT recorders it was slightly higher - 48 kHz. At first, all DACs worked on a “parallel” principle - all 16-bits were “weighted” on an R-2R matrix (a ladder-type resistor circuit).

Example of an R/2R DAC circuit

Connoisseurs know by heart and appreciate brands of chips such as Burr-Brown PCM63 or Philips TDA1541. However, R-2R matrices turned out to be a bit expensive and not very technologically advanced. Precise laser adjustment of all resistance values ​​was required. Otherwise, during operation, inaccurate bit measurement led to a violation of signal linearity.

Therefore, the R-2R was replaced by DACs with 1-bit conversion, called “delta-sigma”. If multibits produced the signal voltage directly, based on all 16-bit data received on the matrix, then in delta-sigma the voltage fluctuated depending on whether the “zero” came to the receiver or the “one”. 1 meant an increase in analog signal voltage, and 0 meant a decrease.

Burr-Brown PCM63 multibit DAC chip

Old audiophiles will remember the musicality of R-2R chips, but there’s nowhere to go. Delta Sigma turned out to be both more practical to set up and cheaper to manufacture. And the quality of the SACD format has proven that 1-bit conversion is excellent at coping with High-End tasks. The SACD sampling frequency is no longer measured in kilohertz, but in megahertz, so the circuit can be used with very simple analog filters.

In classic PCM-based circuits, you still have to filter out quantization noise digitally - there are several of them, and some DAC models provide the ability to choose one of them.

Delta-sigmas themselves progressed towards hybrid circuits, where the stream was processed in cascades, both in 1-bit and parallel circuits. But most importantly, the size of a digital word increased in them, first to 24, and then to 32 bits. In addition, DACs based on field programmable gate arrays (FPGAs) are a promising area, where there are no traditional converters at all.

Modern Mytek Manhattan DAC works with PCM streams 32 bit / 384 kHz, DXD, DSD-DS-DSD256 (11.2 MHz)

Why such an extended bit depth? For authenticity. The professional industry today uses 24-bit recording, which provides a more accurate description of the original signal. As already mentioned, a number of music titles are already available in high resolution format. So you can, of course, listen to the stripped-down version on a CD or MP3, but you must admit, it’s more interesting to get one step closer to the sound engineers who tinkered with your favorite album. And therefore, your DAC must be fully ready to receive high-resolution content - both via USB and other data transfer protocols.