The MSB Ladder DAC is a cutting edge, statement product technology designed by MSB's fanatical engineering team. Unlike virtually all other DACs available today, the MSB DACs do not use any off-the-shelf DAC chips or digital filter chips. Everything is designed and built in house including digital filters, reclocking scheme and other special internal processes that takes advantage of the extraordinary performance of MSB's proprietary hyper bandwidth DAC modules. The MSB DACs offer unprecedented sonic performance and value.
Digital since the mid 80’s
When CDs were designed in the mid 80’s, the Redbook standard was implemented that dictated a certain digital code written in bits consisting of 1’s and 0’s, that could be stored in media like the CD disc. The specific Redbook code would describe an analog music waveform and upon the conversion of the digital bits back to analog, that waveform would be decoded and reproduced. Lacking the inexpensive processing chips that are available today the conversion to digital was done by a resistor ladder. The bits were sent into a simple resistor ladder called an R2R DAC (resistor to resistor Digital To Analog Converter).
For conceptual purposes let's look at the simple R2R ladder diagram. Each bit has a certain voltage or “push” into the ladder. Each bit (pulsed voltage) is a small part of the original analog sine wave signal. The bits are sent into the rungs of the ladder (D0-D7), all bits blending their individual values into the resistor network. Each of these bits are driven into the resistor ladder on the individual rungs contributing its voltage or “push”. These individual bits with their short burst of voltage blend together in the network of resistors resulting in a new continuous voltage which is intended to be the exact copy of the original analog signal that is tapped off at the top and bottom of the ladder. To convert the digital bits to analog precisely, the value of each resistor has to be the exact correct ohm-value requiring very high precision in the manufacture of the ladder. The ladder DAC like the original Philips TDA 1541 was of limited accuracy and very expensive to make. Yet its design is potentially superior because the conversion is done through simple resistors (passive) with no processing. Because the ladder contains only passive resistors, the speed can be extremely high.
Later when the cost of processing chips came down, The Delta Sigma DAC was introduced using a 1 bit method and a form of computer micro processing to decode the digital signal. The Delta sigma is most accurate at the top of its dynamic range but losing resolution at low signal levels (softer sounds). Because the Delta Sigma design converts by processing, extensive analog filtering is required. This additional analog filtering introduces its own set of inaccuracies like phase shift in the output signal.
MSB Sign Magnitude R2R DAC
MSB has always known the ladder was a superior conversion method and introduced the world’s first discrete 24 bit Sign Magnitude R2R Ladder DAC. The term Sign Magnitude describes the special architecture we use that dramatically improves the sound of low level signals. Instead of always starting at the lower limit of the signal and adding voltage to reach the music signal, we start at the midpoint, or zero crossing, where music is quiet, and we either add or subtract voltage to get the required signal. Because this requires a much smaller addition or subtraction on average, it can be done much more accurately.
MSB has designed and built a new proprietary R2R architecture that far exceeds the performance of the original ladder DAC design. The performance of a Ladder DAC is defined by the precision of the resistors. There are hundreds of very expensive aerospace grade resistors on each MSB module producing a DAC with a level of precision that is unheard of. The noise floor (the lowest sound that can reproduced), is much lower than most test systems can even measure. But most important to MUSIC rather than TEST SIGNALS, and very different from Delta Sigma DACs the MSB DAC module are most accurate with signals crossing zero, where music actually exists.
When we talk about digital sample rate we mean the speed of the bits in kHz. CDs are 44.1 kHz (44.1 thousand times per second), and higher resolutions typically go up to 192 kHz with the next generation of hi-res recordings just now available at 384 kHz. MSB DAC modules can operate beyond 5 mHz (5 million times per second), so these modules can receive and reproduce all current formats and conceivable future formats for many years to come. Unlike Delta Sigma DACs the performance of the MSB ladder DAC actually gets better with more bit depth and recording resolution. Low level resolution is recovered to an extraordinary degree. Please see How DACs Work for more information.
MSB Input Processing
The amazing performance of these DACs would be under-utilized if they did not receive the best high-resolution digital signal possible. The front end of MSB's DAC IV series starts at the input receiver where MSB’s proprietary reclocking scheme reduces incoming jitter to under 7 Pico seconds and in some cases less than 2 ps. Since there was no source that could challenge the MSB DAC IV's resolution and jitter, MSB designed an analog-to-digital converter (ADC), that was at least as good as the DAC. This later became a product - the MSB Studio ADC. Then MSB found it was it was necessary to design and build an in-house jitter measurement system to measure the final jitter performance. While some published specs show very low numbers the results are limited by the measurement equipment available.
After the jitter is removed the data signal agrees in time with the clock signal. Then the datastream is sent to MSB’s proprietary Intersample Harshness Correction. This circuit was designed in response to the universal complaints that digital sound was harsh with complex overtones on instruments like multiple horns, multiple violins, massed voices, etc. It is even more harsh when those multiple instruments or voices get louder as the artists express themselves. It was discovered that virtually all factory made discs and recordings we could find violated industry standards resulting in digital “clipping” and the resulting harshness. MSB designed the Intersample harshness correction as a no-compromise correcting process that brings the bit values into the correct signal level freeing up the top end of the signal and associated harmonics to be accurately converted and further improve dynamic range.
The next process is the digital filter necessary to remove artifacts above the audio range that are not related to the analog signal. The implementation of this filter is critical and there are none available off the shelf that come close to MSB's requirements. Digital filters are written in house and installed on a SHARC DSP chip big enough to contain at least 4 of MSB's proprietary filters. These are the fastest DSPs available and run a single-stage 80-bit fixed point FIR Filter, resulting in very fine resolution that MSB's DAC modules can take full advantage of.
So what is a digital filter all about? Because of the very large scale SHARC chips, the digital filter can be written to a much higher resolution (like 32 X) performing a function similar to an upsampler. When done correctly with high accuracy this high resolution digital filter can make an extraordinary improvement in the sound. It is important to understand that the digital bits are nothing more than samples that were taken every so often from the original recording waveform. How frequent these samples were taken determine the original resolution so if they were taken at 44.1 thousand times per second (44.1 kHz), we don’t actually know exactly what the waveform looked like between the samples. This is also true of higher resolutions. This is the one place in the digital conversion where art enters in, and that is in digital filter design. Lets look at the process. We start out with a continuous analog stream at the studio. Every so often we record a single voltage point on the analog wave form. These points enter our DAC and our job is to figure out what was between the points.
So let's take a simple example. Here are three dots. . . . Now they can be connected a lot of different ways, with a straight line, with a square wave, or with a sign wave. Each will sound very different but all are technically correct possible interpretations of what might have been present between the original data points. This is what a digital filter does. It tries to make an intelligent guess what the original music looked like. One of the digital filter techniques looks way back in the past, and way forward in the future, and uses this data, plus what we know about the nature of music to make a better prediction about what was between the data points. This is where the art and many years of knowledge and experience come into play making the best digital filters possible. The “algorithms” are the predictive math that is based on what came before the sample we are looking at and what came after the sample we are looking at. The algorithms are designed to go back and go forward thousands of samples just to decide what was most likely in between the two original bits we are trying to fill in at that particular time. It is easy to see why such large extensive high-speed processing is needed. We fine tune these algorithms by testing, and then listening, to make a filter that guesses better. We have tried hundred, if not thousands before arriving at our current filter suite.
The whole rest of the audio chain just applies good engineering to be as accurate and noise free as possible. But in this case we are artists, trying to paint the best interpretation of the same data points all our competitors are also looking at. That's what makes our product sound different and why we offer so many filters and upsamplers. None are perfectly right or wrong, they are just different interpretations of the same data, that may match the studio conversion better for one recording than another. This very high-density data is easily resolved and converted by MSB's Platinum, Signature or Diamond DAC modules since they are capable of converting at more than 5 mHz.
You will remember from the earlier discussion about the original Philips ladder DACs that the bits have a voltage and that voltage blends into the resistor ladder to make the analog signal, the final continuous voltage that is now the analog waveform (music). A happy result of the MSB ladder DAC design is the output voltage of the DAC is in excess of what is needed to drive an amplifier. Therefore all that is required to achieve a very high performance volume control is a stepped resistor attenuator. After the stepped attenuator there is a single obsessively designed output buffer to be sure the impedance at the output jacks stays very low. The result is that this DAC is able to drive any amplifier no matter how difficult.
The performance of DAC IV is amazing. Immediately you'll notice the beauty of the voices and the lack of congestion and harshness that often accompanies massed voices and massed instruments with complex harmonics like violins and saxophones. This new level of clarity is also noticeable in instruments of all frequencies. This DAC also dramatically increases the resolution of the performance. Fine detail that was once hidden and "smeared" in the background is now clear and obvious. There is a perception that each fine harmonic is individually revealed giving new definition and separation to the instruments. The sense of space where the recording was done is intricately revealed with wider and deeper soundstage. Each instrument is more singly placed and positioned. Our engineering and listening team have always noticed that when genuine fine detail is further revealed (without harshness), there always seems to be an increase in the sense of space. This makes sense because spatial cues are the original sound of the voice or instrument with the reflections from the walls of the recording space added, and are therefore complex harmonics themselves.
The dynamics are significantly improved giving the drummer's expression a convincing "hit" that cannot be ignored. Bass attack is convincing and scaled to what the artist intended. This new level of dynamic reproduction is a critical ingredient in convincing us we are listening to the live performance, and drawing us deeply into the music. Once it is heard, it is hard to live without.
One theory of what is going on with system matching
Now that we have some experience with the sound quality of the DAC IV including comments from around the world, we feel that the DAC IV achieves a new paradigm in sound performance. We also have realized it has had a profound effect on system matching. It is our opinion that for over 25 years now, audiophiles (ourselves included), have been tuning their systems to reduce the harshness and grain that they hear in their systems. Often in addition to the digital front end, the preamplifier, amplifier, cables, or wall power are blamed. While any of these components can show an improvement and can have better performance on their own, we also suspect that certain components can sonically "slow" the signal making digital more tolerable with decreased harshness and increased clarity. Because digital is math, and crimes (errors) in that math can have incredible speeds and rise times these can be transferred to the analog signal. These errors may not pass through certain components downstream like colored tube or solid state equipment, certain cable designs, or even some speaker crossovers with their necessary inductance and capacitance. So if digital has been harsh all these years and certain components "slow" the signal and have a tendency to subdue the really fast and sharp rise times and let the more rounded sine waves through, maybe this sounds more like the original analog wave form and the natural sound of the instruments. If this is true then it is easy to see how complex it would be to correlate a good sounding result with more accurate components vs colored components.
So if previous digital has been harsh and we have been tuning our systems for it, and the DAC IV is really a new level of fine harmonic detail lacking harshness, then the front end may well be the most important component in the system. As audiophiles we have found that the introduction of the DAC IV in an otherwise known system changes the game. As an example, certain components like a well loved preamp that was once a stable and treasured part of the system can be found to be in the way of the sound. We have also noticed that changes downstream are now laid clear for what they are - good or bad. There is little or no "waffling" about how valuable a change is like: Did the cymbal have a little more "air" with this amplifier vs that amplifier, or cable, or speaker, or wall power treatment? Not only is it clear that a change downstream is good or bad but it is also clear why. The soundstage may be better, or better pace and rhythm, or natural female vocals, etc. As audiophiles we hate to think about all the sideways moves we made over the years that were expensive and time consuming. We are thrilled to know that a better source like the DAC IV is true to the music, making system matching an easier and more efficient process.
We invite the user to peel back the layers of correction and listen to the DAC IV with internal volume control in the simplest of configurations. With a good bit-perfect source, and fast and accurate amplifier and your favorate speakers. We are confident you will hear a new level of realism never before experienced with your system.