Balanced-Ouput DACs and Tubes and Aikido
Back to DACs. Last time we looked at this topic, we saw tube circuits picking up their input signal from DACs with a single output, which was either a voltage or a current output. So far, everything has been fa however, when we come to DACs with balanced outputs, we face a few twists. For example, most CD players and stand-alone DAC units that sport balanced-output DACs are not balanced themselves at their outputs, with a single RCA jack per channel as an output. Here is another oddity: the output of a non-balanced DAC is usually referenced to ground, with an output that expects to see 0V. On the other hand, balanced outputs are usually referenced to some positive voltage, say 2.5V.
So, the first question is: Why use balanced-output DACs when the final output is unbalanced? The answer is that we can get better performance at less cost. When the DAC's two phases of output signal are summed together, the summing circuit&s intrinsic common-mode rejection ration (CMRR) will ignore the noise that is common to each balance output, amplifying only what is not common in its single-ended output, which affords an improved signal-to-noise ratio. In other words, the power-supply noise falls out of the mix, so a simpler and cheaper DAC power supply is allowable.
CMRR and differential amplifiers are seldom
mentioned in tube circles. The following illustrations expose a differential amplifier&s functioning.
Now, we can see how the differential amplifier, by ignoring what is common and amplifying what is different between balanced inputs, can produce a clean output waveform from two dirty signals. (Note that when both inputs see the same voltage, the output falls to zero.)
The next question is: Why do so many new DACs have outputs that are not at the ground potential? Historically, the trajectory that DACs have been on started with many pins and many support devices and many supply voltages to the desired end of only a few pins and only one power supply voltage. The reasoning is easy to decipher&it&s both easier and cheaper that way.
The makers of digital-audio gear complained of the cost of all the required prop-up components, such as decoupling capacitors and trimming potentiometers, and excessive complexity (they had been spoiled by OpAmps, which are cheap and easy to use); and the makers of DACs wearied of the many user-support headaches, as their DACs failed to deliver the advertised performance due to their customers' lack of understanding and sloppy construction practices. In other words, no one was happy. Thus, we are getting DACs that are simpler, cheaper, and easier to implement.
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In the above DAC footprints, we can see the move to smaller and simpler. (Each of the pins labeled &DECOU& and pins 16 and 17 require
and the DAC requires three power supply voltages, +5V, -5V, and +15V.) However, a single power supply voltage of 5V cannot accommodate a ground-voltage-referenced output, as there is no negative-power-supply voltage to support a current source to pull the output down to ground level. (If the output were inductively loaded, however, such a DAC w but who& other than an audiophile &wants an expensive, heavy, hum-attracting choke in his or her iPod or portable CD player?) Thus, many of the new, streamlined DACs have an output referenced, not to ground, but instead at B+ voltage divided by two (or some value close to two). For example, if the DAC&s power supply voltage is 5V, then its signal outputs will be at +2.5V. So is a coupling capacitor
needed to block this large DC offset? No, because the balance-to-unbalanced circuit will drop out the 2.5V DC offset along with the noise, as the shared DC offset voltage is effectively a common mode signal.
So where the old DACs were complex, requiring many clean power-supply voltages and many bypassing capacitors, the new breed of easy-to-use DACs requires only one power supply voltage, and it need not be perfectly clean for a fairly decent PSRR.
So how do we use the balanced-output DAC with a vacuum tube output stage? Let's pull back and review. The balanced-output DACs, with internal I-to-V converters and voltage outputs, put out a good amount of signal swing, easily enough to drive the average power amplifier directly. If you need more gain, however, the design of the the tube circuit will have to be fairly complex, as it will
have to comprise gain and buffering stages. But if the tube stage does not have to drive a low-impedance load, the following simple circuits will work quite well, converting balaced input into an unbalanced output, while providing a CMRR roughly equal to 1/mu.
In both circuits, the addition of resistor R in the cathode circuit, allows the tube circuit to have its effective reference voltage upped to 2.5V.
In other words, without the DAC, the tube circuit would bias up to where the connection between RK and R would naturally sit& at 2.5V at idle. Of course, we are not limited to the two above tube circuits. For example, the Aikido amplifier can be used just as easily:
The input stage of the Aikido is effectively lifted 2.5V off ground by the addition of the extra resistor. By switching to the Aikido we gain all the benefits that the Aikido offers, such as low output impedance, low distortion, and a great PSRR. The only obvious problem is that of too much gain (the previous two circuits also suffer from this), as the balanced signal from the DAC will result in twice the gain we would have gotten from a comparable unbalanced signal source. In the case of the Aikido, the calculation is an easy one: gain roughly equals amplification factor of the input tube, for example, 20 for the 6SN7 and 33 for the 6922.
Another less obvious problem that we might encounter with all these circuits is that the wimpy internal OpAmps in the DAC might not be able to drive such a low-impedance load. Which low-impedance load? The non-inverting outputs (in these schematics at least) have the low-valued resistor that bridges the cathode resistor to ground. This resistor is also in parallel with the low output impedance presented by the bottom triode's cathode in series with the cathode resistor. What can be done? The following circuit unloads the internal OpAmps by the current gain of the transistors used (typically, well over 100).
Oh my, how amazingly creepy: two transistors! Try as I may, I keep forgetting that if a nervous audiophile cannot see a transistor, such as the several hundred in the DAC, the transistors do not exist. Okay now, is there a circuit that isn't so creepy and doesn't have excessive voltage gain and doesn't burden the internal OpAmps within the DACs? (This would be much more dramatic with a beautiful, scantly-clad, showgirl assistance and a shower of sparks...)
What we are doing here is throwing away much of the DAC's output signal, while providing a higher-impedance load to the DAC, and while retaining the high CMRR. Extra touches can take the form of high-frequency-shunting capacitors added to filter away some of the high-frequency noise from the DAC.
(The same buffer/attenuator circuit can be used with the first two circuits shown, the grounded-cathode and SRPP amplifiers.)
On the other hand, if no gain is required, then a much simpler circuit can be used. (Remember, no gain, no pain.)
has been covered here before, so only a quick recap will be given. This circuit accepts a balanced input, which it converts to an unbalanced, single-ended output (the circuit itself is push-pull in functioning). It offers low distortion and low output impedance and, most importantly in this discussion, a good CMRR figure.
The Broskie CF can accommodate a low-pass filter by adding two capacitors, C1 and C2. These capacitors limit the high-frequency bandwidth, which will help filter away high-frequency DAC generated noise. (I would place the transition frequency at 30kHz with 500pF capacitors.)
Expect more balanced-output DAC circuits, but with differential amplifiers at their core. (I figure this blog entry is the most that anyone reasonably can digest before rushing off to work or school. I keep striving for &bite sized,& but big lunches usually seem to win out. Today I hope I have managed to keep it digestible.)
Kit User Guide PDFs
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> ICM7357BQG (IC MICROSYSTEMS SDN BHD) Quad 12/10/8-Bit Voltage Output DACs with Serial Interface and Adjustable Output Gain PDF资料下载
Quad 12/10/8-Bit Voltage Output DACs with Serial Interface and Adjustable Output GainRev. A8 ICmic reserves the right to change the specifications without prior notice. 8 IC MICROSYSTEMSICmicDETAILED DESCRIPTION The ICM7377B is a 12-bit voltage output quad DAC. The ICM7357B is the 10-bit version of this family and the ICM7337B is the 8-bit version.
This family of DACs employs a resistor string architecture guaranteeing monotonic behavior. There is a 1.25V onboard reference and an operating supply range of 2.7V to 5.5V.
Reference Input There are two reference inputs that can be driven from ground to VDD–1.5V. Determine the output voltage using the following equation: VOUT = VREF x (D / (2n)) Where D is the numeric value of DAC’s decimal input code, VREF is the reference voltage and n is number of bits, i.e. 12 for ICM7377B, 10 for ICM7357B and 8 for ICM7337B. Reference Output The reference output is nominally 1.25V and is brought out to a separate pin and can be used to drive external loads. The outputs will nominally swing from 0 to 2.5V. Output Amplifier The Quad DAC has 4 output amplifiers with a wide output swing. The actual swing of the output amplifiers will be limited by offset error and gain error. See the Applications Information Section for a more detailed discussion.
The 4 output amplifier’s inverting input of 4 DACs are available to the user, allowing force and sense capability for remote sensing and specific gain adjustment. The unity gain can be provided by connecting the inverting input to the output. The output amplifier can drive a load of 2.0 k to VDD or GND in parallel with a 500 pF load capacitance. The output amplifier has a full-scale typical settling time of 8 μs and it dissipates about 100 μA with a 3V supply voltage. Serial Interface and Input Logic This quad DAC family uses a standard 3-wire connection compatible with SPI/QSPI and Microwire interfaces. Data is loaded in 16-bit words which consist of 4 address and control bits (MSBs) followed by 12 bits of data (see table 1). The ICM7357 has the last 2 LSBs as don’t care and the ICM7337 has the last 4 LSBs as don’t care. The DAC is double buffered with an input latch and a DAC latch. Serial Data Input SDI (Serial Data Input) pin is the data input pin for All DACs. Data is clocked in on the rising edge of SCK which has a Schmitt trigger internally to allow for noise immunity on the SCK pin. This specially eases the use for opto-coupled interfaces.
The Chip Select pin which is the 8th pin of 20 QSOP package is active low. This pin must be low when data is being clocked into the part. After the 16 clock pulse the Chip Select pin must be pulled high (level-triggered) for the data to be transferred to an input bank of latches. This pin also disables the SCK pin internally when pulled high and the SCK pin must be low before this pin is pulled back low. As the Chip Select pin is pulled high the shift register contents are transferred to a bank of 16 latches (see Figure 2.). The 4 bit control word (C3~C0) is then decoded and the DAC is updated or loaded depending on the control word (see Table 1).
The DAC has a double-buffered input with an input latch and a DAC latch. The DAC output will swing to its new value when data is loaded into the DAC latch. The user has three options: loading only the input latch, updating the DAC with data previously loaded into the input latch or loading the input latch and updating the DAC at the same time with a new code.
Serial Data Output SDO (Serial Data Output) is the internal shift register’s output. This pin can be used as the data output pin for Daisy-Chaining and data readback. And it is compatible with SPI/QSPI and Microwire interfaces. Power-On Reset There is a power-on reset on board that will clear the contents of all the latches to all 0s on power-up and the DAC voltage output will go to ground.
APPLICATIONS INFORMATION Power Supply Bypassing and Layout Considerations As in any precision circuit, careful consideration has to be given to layout of the supply and ground. The return path from the GND to the supply ground should be short with low impedance. Using a ground plane would be ideal. The supply should have some bypassing on it. A 10 μF tantalum capacitor in parallel with a 0.1 μF ceramic with a low ESR can be used. Ideally these would be placed as close as possible to the device. Avoid crossing digital and analog signals, specially the reference, or running them close to each other. Output Swing Limitations The ideal rail-to-rail DAC would swing from GND to VDD.However, offset and gain error limit this ability. Figure 6 illustrates how a negative offset error will affect the output. The output will limit close to ground since this is single supply part, resulting in a dead-band area. As a larger input is loaded into the DAC the output will eventually rise above ground. This is why the linearity is specified for a starting code greater than zero.
Figure 7 illustrates how a gain error or positive offset error will affect the output when it is close to VDD. A positive gain error or positive offset will cause the output to be limited to the positive supply voltage resulting in a dead-band of codes close to full-scale.
Quad 12/10/8-Bit Voltage Output DACs with Serial Interface and Adjustable Output Gain
Quad 12/10/8-Bit Voltage Output DACs with Serial Interface and Adjustable Output Gain
Quad 12/10/8-Bit Voltage Output DACs with Serial Interface and Adjustable Output Gain
DUAL 12/10/8-BIT VOLTAGE-OUTPUT DACS
DUAL 12/10/8-BIT VOLTAGE-OUTPUT DACS
相关代理商/技术参数
制造商:ICMIC 制造商全称:IC MICROSYSTEMS 功能描述:SINGLE 12/10/8-BIT VOLTAGE-OUTPUT DACS
制造商:ICMIC 制造商全称:IC MICROSYSTEMS 功能描述:SINGLE 12/10/8-BIT VOLTAGE-OUTPUT DACS
制造商:ICMIC 制造商全称:IC MICROSYSTEMS 功能描述:SINGLE 12/10/8-BIT VOLTAGE-OUTPUT DACS
制造商:ICMIC 制造商全称:IC MICROSYSTEMS 功能描述:SINGLE 12/10/8-BIT VOLTAGE-OUTPUT DACS
制造商:ICMIC 制造商全称:IC MICROSYSTEMS 功能描述:SINGLE 12/10/8-BIT VOLTAGE-OUTPUT DACS
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添加更多采购How DACs WorkHow DACs Work
Let's start at the beginning.
Digital music is an easily transported intermediate form between an analog original and an analog copy.
An ideal system creates a copy at the end that is identical to the original.
That has yet to happen, but over the last 20 years we are getting closer and closer.
The two critical components of this process are the Analog-to-Digital Converter (ADC) at the studio and the Digital-to-Analog Converter (DAC) in your listening room.
Let's start with a look at a diagram of the ADC process.
The Studio Side - ADC
ADCs work by repeatedly measuring the amplitude (volume) of an incoming electrical pressure sound wave (an electrical voltage), and outputting these measurements as a long list of binary bytes. In this way, a mathematical &picture& of the shape of the wave is created. Do not worry about bits and bytes.
For our purposes, these are simply numbers, and the &bit depth& is just an indication of the accuracy of the number, kind of like 6.24 is more accurate than 6.2.
So what is this waveform we are recording and trying to recreate?
The single cycle wave form in our example is the analog or copied result of all the frequencies from all the instruments that occurred in the air in the recording studio and naturally combined in the air and came piling into the single point microphone diaphragm in a certain order yielding one analog sound naturally produced as our eardrum would hear it. This uncountably large number of frequencies from all the instruments and their harmonics and the resulting room reflections (sound stage cues), naturally combined in the air and the naturally &coded& complex waveform that is the original truth of the music at that moment is what we are trying to copy accurately.
Join the dot pictures
Remember join-the-dot pictures? To produce a good image you must have sufficient dots to capture the detail of the shape AND the dots must be positioned accurately.
Quality in a join-the-dots picture depends on ...
Number of dots (Horizontal X axis) which has to do with time.
Accuracy of the positioning of the dots (Vertical Y axis) which has to do with amplitude or loudness at that instant in time.
Number of Dots = Sample Rate
The number of dots is the sample rate, or the number of samples taken per second.
The more samples taken, the easier it is to recreate the original.
But these samples need to be made accurately.
The jitter of the clock is the variation in time of each cycle.
Shifting the time of the sample records the wrong level.
Therefore the accuracy of the digital clock, which governs when samples occur, is paramount. If the clock is not accurate, jitter will occur and the audio quality will suffer.
Accurate Measurement of Amplitude
Measuring amplitude is like measuring with an elastic ruler.
If the smallest divisions on the ruler are 1 cm than each measurement must be rounded to the nearest cm.
The accuracy of the ruler is also important.
Any stretch of the ruler will give a false reading. So why not use much smaller divisions?
For a CD, these are defined by the Redbook standards and cannot be changed.
Bit Depth (the resolution of the measurement scale). Bit depth in images determines the number of possible colors a pixel can be.
In audio it is the scale of the ruler measured in number of bits.
A 24 bit recording has a finer scale than a 16 bit recording.
Measurement Accuracy (how accurately the amplitude is measured). Where the bit depth determines the resolution or scale used to measure, accuracy is like the precision or stretch of the ruler.
This is generally a hardware function, like the precision of a resistor, not a timing issue.
What's Lost is Lost
At this point we have lost two big things.
We have lost the exact analog level of each sample.
Because of the limited resolution of each bit step, we have lost the exact amplitude of each point.
The amount a sample measurement has been rounded up or down is known as the quantization error and produces quantization distortion. At loud signal levels quantization errors manifests themselves as noise (similar to analog noise), but at low signal levels they can manifest themselves as unwanted audible distortion.
The other thing that is lost is the shape of the curve between each point.
It is gone forever, again a Redbook limitation.
But as you will see, we work very hard in the DAC to try to get as much of this missing information back.
Now its Digital and Portable.
The sample rate was defined and the ADC has done its job and locked into the data all its error, but the data is now just a large file of numbers, and these can be changed into innumerable formats and transported around the world and is finally presented to your listening room, to your DAC.
Bit Perfect
If the integrity of the file has been maintained, the exact same numerical values that were generated in the ADC are presented to your DAC.
If it is, the data is said to be Bit-Perfect.
Sample Rate Conversion
Unfortunately many times
a computer or product which works with digital music from more than one source, finds it much easier to process, if all music is converted into the same sample frequency and same bit depth.
This all too common process double the original ADC errors, by resampling the data a second time.
Computers normally always output audio at the sample rate and frequency the user specifies, regardless of its original sample rate.
The Playback Side - DAC
A DAC works by reading the digital data in a file and attempting to recreate a copy of the original analog signal recorded.
We face many of the same problems faced in the recording process.
Basically a DAC as three jobs:
Sample Amplitude The DAC creates an analog voltage equivalent to the numeric value read from the file.
A DACs ability to do this depends on the type of DAC it is and its accuracy.
More about this later.
Sample Timing The DAC needs to create this analog voltage not only at the right level, but at the right time.
The accuracy of the clock and the amount of clock jitter, used to create these voltage levels is vitally important.
Fill In The Blanks The basic DAC creates an analog voltage equivalent to the numeric value read from the file.
But we know that there is data missing between each sample, and that the original level may have been a little more or less than the level recorded (quantization error).
This is where the science ends and the art begins as we try to guess what the errors were and what is missing between the blanks.
Digital Filter - Fills in the Blanks
The digital filter is the computer algorithm that looks at the digital audio in the past, and the digital audio in the future and tries to figure out what was going on with the music, and tries to shape the final analog output to match as closely as possible that original waveform, now missing for ever.
At this point you may have some questions:
How do you look at the future music data? The only way to see the future, is to delay the output.
So our digital filter is actually way ahead of the output, and why DACs create audio delay.
If the original is missing, how do you know what kind of filtering is right? Even though the original is missing, our sensitive ears and memory know what the human voice and various instruments sounds like.
We listen to each new algorithm we write and ask ourselves if the result is more life-like or not.
This is the art of the process.
We see, knowing the nature of music, if we can fill in the blanks perfectly.
This method would not work for random data, but sound generating instruments follow natural rules (because the strings on a guitar have a certain weight and tension and therefore must have a certain resonance, and the overtones that give that guitar it's character all follow the natural rules ).
Sample Timing - The Clock
We talk a lot about clocks.
We are not talking about devices to tell us what time it is,
but a device that outputs a pulse or cycle at a fixed interval or time.
If you counted them up you could tell time, but we use them to create an absolute reference which is the same at the recording and playback ends.
Normally DACs do not contain clocks.
The clock is associated with the device that reads the digital media.
So CD transports have clocks, DVD players have clocks, Computers have clocks.
The clock is sent to the DAC along with the digital data.
This method allowed DACs to be very low cost but this approach severely handicaps the DACs performance.
Unfortunately today all but a handful of high-end DACs still follow this method.
Our Solution - Reclocking - The MSB approach to this problem is to accumulate data from the source in a memory, and then read it from the memory using a super-accurate low-jitter clock, located right in the DAC.
This feature is included in all MSB DACs.
Even Better - PRO I2S with Master Clock Output - Although reclocking is fantastic, we still have data coming in from an outside source at a different frequency from our internal clock, being clocked by a transport or computer.
Having these two slightly different clocks running in the same box is a cause of noise.
When we feed our DAC clock back to the source, and synchronize it with the DAC, we eliminate this source of noise and get even better timing accuracy.
Sample Amplitude - The DAC Accuracy
This critical parameter is equally important.
Just as we can hear tiny variations in the clock, we can hear errors in amplitude (moment to moment loudness).
The accuracy of a DAC depends on the quality of the DAC components as well as the DAC architecture used.
Almost all DACs produced today are Delta Sigma because of the amazing average accuracy obtained over many samples.
MSB uses a Ladder DAC, able to achieve similar accuracies, but in every sample and at a greater cost.
Types of DACs and How they Work
A DAC is a circuit that converts digital measures of audio
amplitude in discrete steps into a continuous analog electrical equivalent of
the sound to be reproduced.
The amplitude is a digital number (like a 16 bit
word) and the steps occur based on the sampling rate (like 44,100 times per
This process is very much like an endless conveyer belt with empty
one gallon jugs on it, moving by a filling station.
The size of the jug is
fixed, the rate they pass by is determined by the sampling rate.
The goal of
the DAC is to fill each jug to exactly the right level specified by the music.
There are three techniques use Delta Sigma, Ladder, and
the MSB Sign Magnitude Ladder.
Delta Sigma DACs
(Single Bit)
Each sample or jug is filled to the right level with many
measuring cups, poured to reach the target level.
A &one bit& measuring cup is either full or empty. With 64
times oversampling, the cup is only 1/64th the volume of the jug.
This rather
crude cup does not come close to being accurate enough.
The cup would need to
1/16,777,216th the volume of the jug to be accurate.
Line speed is like Sampling Rate (48,000 per second)
By keeping a running tally of the error and by going a little
over and a little under in many many samples, a very accurate average is
reached, but only with an aggressive filter applied to the output.
Ladder DAC
A ladder DAC is different in that instead of a single
measuring cup (or bit), a whole array of cups are available, from very small to
very large.
Any combination of cups can be used to fill each jug to
exactly the right level.
No filtering is required but the accuracy is defined by the
cumulative error of all the cups used.
MSB Sign Magnitude Ladder DAC
The MSB Sign Magnitude Ladder DAC is like a Ladder DAC but
refined in two ways.
Because the jars are typically 1/2 full when finished, we
start with very accurately 1/2 filled containers instead of empty ones.
there, we again use the wide range of measuring cups to either add or take away
from each jar.
Again each jar is filled exactly so no filtering is needed,
but because we only had to add or subtract a little bit our accuracy is higher.
Because our hearing is most sensitive to low level sound (sound near the 1/2
full zero crossing) our DAC is most accurate near 1/2 full where we use the
smallest and most accurate measuring cups.
What about Upsampling?
Synchronous Upsampling like MSB does
just means adding more
jars between the existing jars and moving them faster
down the line.
By looking at many jars before and after the new empty one, we
calculate how full to make it.
The result is smaller steps as shown in the
This is very significant when used with a Delta Sigma DAC but less so
with a Ladder DAC.
The Final Word
The original way Sony and Philips converted digital was a ladder DAC, just a handful of resistors that when connected looked like a ladder (hence the name). It was very difficult for Philips to build because it was hard to get the resistors accurate enough, so the original 1541 DAC
was inaccurate. Yet it sounded better than the next generation method, the Delta Sigma DAC. (BTW, the 1541 Crown DAC is still sought after today by boutique companies and going for up to $200 each compared to their original cost of $12. It is nice as MSB has a drawer full of them!) The ladder DAC is technically, attractive because it is like a passive process (zero processing of any kind, yes I know the conversion itself is a process), meaning there are no active sampling and averaging processing, only simple resistors that have no speed limit. The MSB ladder is a refinement of the original design. The resistor network is faster - more than 5 MHz.
It has a noise floor of 160db - keep in mind that each 3 db is twice as loud. And it is far more accurate with the highest precision resistors made in the world today.
So what is the MSB approach?
Start with native resolution.
Reading the bits natively has been proven by ear to always be best (the Data CD)
Digital Process at high resolution
We process that data with our own digital filter, at 80 bit resolution with code written in-house to optimize the incredible capability of the MSB DAC modules.
Reclock with an accurate low jitter clock.
We reclock that data with the lowest jitter in the industry.
Convert to analog with a sign magnitude ladder DAC. MSB offers three levels of DAC precision, and the output of the DAC modules is very robust and has more power than is necessary to drive an amplifier so we can offer a very neutral volume control internally by simply using resistors to attenuate the output voltage.
Unlike virtually all other digital manufacturers, MSB is a truly unique company designing and building all processes and hardware in-house so we are never limited by components, hardware, chipsets or the software processes of DAC chip manufacturers.
We invite you to listen to MSB and compare the sound to the best you can find at any price. Please contact us for an in-home demo. You will never think about digital playback the same way again.