Ultimate-Preamplifier 2 **
A truly no compromise high-end DSP based digital preamplifier for stereo or custom multi-channel applications !
The Ultimate-Preamplifier 2 (UP2) is a fully stand-alone digital preamplifier aimed at high-end audiophile Stereo and Multi-Channel audio applications. At its heart It contains an Analog-Devices SHARC DSP which can easily be configured with Audio Weaver™ from DSP Concepts.
Along with the ESStech ES9038PRO Sabre DAC, a wide variety of digital and analog audio inputs and outputs you now have the ultimate DSP processing platform to build your next ultimate active speakers ! Stay tuned for our companion multi-channel power amp specifically tailored toward active speaker environments !
- Esstech ES9038PRO Sabre Reference Audiophile DAC. With ESS patented 32-bit Hyperstream™ DAC architecture and Time Domain Jitter Eliminator, the SABRE32 Reference Stereo DAC delivers an unprecedented DNR of up to 135dB and THD+N of -120dB, the industry’s highest performance level that will satisfy the most demanding audio enthusiasts.
- DAC-IV stages completely redesigned using high current differential, ultra wide bandwidth amplifiers with purpose built optimal bias servo control for each DAC output to establish the optimum operating point for each DAC,
- Ultra low-noise analog DAC supplies using the Esstech ES9311Q regulator
- Femto clock with ultra-low phase noise performance, with dedicated low noise regulator circuit and clock distribution network designed to mitigate jitter by optimizing PCB layout with the shortest possible traces and keeping all peripherals on the same board,
- Dedicated 2 channel USB class-2 interface can handle PCM processing up to 192KHz
- Sabre DAC Master mode available for 2 channel and 8 channel preamp modes
- Perfect Link is available to link and synchronize multiple preamps together for more channel and DSP capability without any additional delay between preamps. Timing delay is within one MCLK period or 20 nS.
- Our own proprietary in-house Adaptive Intersample-Over filter makes sure that the asynchronous sample rate converter does not saturate or clip when it is dealing with differing sample rates !
- Use of MELF resistors where appropriate for lower noise and distortion
- Low noise LDO regulators for the main analog supplies for the quietest operation
- Double the filter capacitance in the power supply with double pole RC filtering to reduce supply ripple noise from the mains supply
- Mains input conditioning including DC blocker for optimum operating point of the power transformer which improves long term reliability
Features in Detail
The Ultimate-Preamp can function as a stand-alone stereo or multi-channel preamp which can process both analog or digital signals including signals from a magnetic cartridge. Precision RIAA equalization is done in the digital domain by an Analog Devices 4th generation SHARC DSP within a 0.01dB tolerance of the inverse RIAA curve and without the problems of a passive design. Unlike analog or passive designs, multiple RIAA curves can be easily accommodated simply by changing the digital filter coefficients whilst still maintaining precise conformance with the standards. Other digital sources can be processed from any one of six galvanically isolated S/PDIF sources and one galvanically isolated I2S interface. In addition, through its galvanically isolated USB interface it can also process PCM up to 192KHz and DSD 64/128/256 source material.
Secondly with its 8 full channels of DSP processing the Ultimate Preamp can form the basis of a stereo 2/3/4 way active stereo speaker system (or 8-way mono) with precision crossover functionality using the full 80 bits fixed point precision or 40 bits floating point precision of an Analog Devices 4th generation SHARC DSP. The Ultimate-Preamp is in a class of its own. With the Link Port adaptor, it is possible to chain two or more boards together for more channels and/or processing power with no latency issues. This allows self-contained fully active speakers to be designed.
Lastly the Ultimate-Preamp can function as a high quality stand-alone stereo DAC. With the industries best DAC hardware and the ability to process digital signals in both PCM and DSD formats from a number of sources including S/PDIF, Toslink, USB-2 audio and I2S. In this case the DSP is not used to process the audio. It is only used for the house keeping functions such as programming the DAC, display information on the LCD display etc.
In all three operating modes, bit-perfect volume and channel level controls are accomplished with true 32-bit precision irrespective of the source and mode of operation. This equals or betters the performance of the best analog volume controls but with the precision, reliability and noise free operation from digital signal processing.
You’ve spent a long time and money building up your vinyl collection so you are going to want to preserve their pristine condition as early as possible. It has been said that the tip pressure of a stylus in a groove can reach as high as 14 tons per square inch which results in a contact temperature as high as 300 degrees Fahrenheit. The friction between stylus and the record can heat the vinyl locally to temperatures near the melting point so some argue that there is degradation of the vinyl after each playback. In fact, an old advert from Goldring many years ago made a point that a typical LP record is only good for 200 plays. Then compound that with wear and tear of a stylus along with air born contaminants finding their way into the grooves, it is imperative that some way of archiving the vinyl playback is needed as early as possible.
The Ultimate Preamp can not only playback vinyl with its built in precision Analog-to-Digital Converter and DSP based RIAA playback equalization but can also archive this playback via an S/PDIF or Toslink output or with the Ultimate Preamp Plus – directly to an external PC equipped with a USB interface thus allowing you to archive and preserve your vinyl collection for posterity sake.
For the first time a multi-way DSP based crossover can accept and directly process Direct Stream Digital (DSD) Audio through its own proprietary on-board DSD processing engine without any software overheads or having to convert it to an analog signal first to be resampled by an Analog to Digital converter. DSD is usually neglected on DSP board offerings simply because to do it in software by the DSP would consume a large proportion of the DSP resources or in many cases require an additional DSP ! We’ve solved this problem by building a DSD decoding engine from the ground up into the on-board custom logic hardware. From DSD-64 all the way up to DSD-256 the on-board DSP decoding engine can process it with ease whilst the DSP is free to do other processing.
The heart of any digital audio system is a precision low phase noise clock generator coupled with a clock distribution network and in this instance we didn’t disappoint. We used the best low jitter clock oscillators from NRZ along with low noise power supplies so that clock purity is maintained. The clock signal from the oscillator is then buffered through a dedicated clock distribution chip and routed to the other devices on the board using impedance controlled traces and the shortest possible trace length thus adding negligible jitter to the clock signals.
In addition to the DSP we have incorporated two large gate count custom logic devices to do tasks that would be difficult if not impossible to achieve in the DSP. Code named ‘Rubidium’ and ‘Iridium’ after their respective chemical elements. Like the use of its chemical counterpart in precision atomic clocks the Rubidium custom chip provides precision timing signals to the DSP, DAC and ADC as well as support for the processing of native DSD data streams which would otherwise consume a large portion of the DSP capacity that would no longer be available to other audio processing needs. The Iridium device performs ancillary functions such as glue logic and support for the LCD display without the need for a second microcontroller.
To communicate with the user, we didn’t just leave them in the dark with a non-existent user interface or one that exclusively relies on lugging a PC around! We’ve incorporated the necessary hardware to build an intuitive user interface. A fast pixel rendering, high resolution full color graphic LCD display coupled with navigation controls provides a context sensitive menu system. For user controls we added a rotary control which doubles as a master volume and menu adjustment control. Along with five navigation buttons the menu is both easy and intuitive to navigate. Seven dimmable status LEDS provide real time status of various conditions. Finally, an infrared remote control interface which is compatible with a standard Apple remote essentially duplicates the functionality of the navigation buttons.
We spent a lot of time perfecting the hardware and firmware for system integration applications. For the ultimate in standalone active speakers each of which would have its own DSP processing hardware we provide the means of linking speakers together so that each board operates in perfect word sync right down to the resolution of each boards on-board master reference clock period of 20 nS ! Typically, other DSP systems ignore this often-overlooked issue and run their boards asynchronously to each other so it’s not uncommon for them to drift out of sync by a few word clocks at best simply because there is no means of synchronizing them together. Worse still they can slip out of sync by up to one hundred word clocks per second depending on the tolerance of each boards master clock. To compound the issue is how to split the incoming digital audio signal from a single USB or S/PDIF connector and make it available to multiple DSP boards?
Our proprietary on board Perfect-Link hardware enables multiple DSP boards to be linked together using a simple connection whilst still being clocked from their own on-board low phase noise master clocks. This results in zero-phase shift between the DAC’s on each separate board as though all of the DAC’s were located on the one single board and being clocked from the same low phase noise master clock!
For standalone speaker applications all that is required to link the speakers is a Simple-Link cable which can be obtained in lengths of up to 5 meters! No more messy and expensive speaker cables running around the floor! Whether you are building a simple active two-way speaker system or multi-way behemoth each with its own self-contained DSP and amplifiers it is mandatory to maintain perfect synchronization otherwise there will be undesirable phase delays between the speakers and that just isn’t considered hi-end these days! Looking at it another way, would you tolerate a standalone DAC which had considerable delay between left and right channels? No you would think it had a design flaw and yet many people operate DSP hardware in this way!
For any DAC or SRC that up-samples a low rate digital audio-stream to a higher rate audio-stream, it is inevitable that some of the intermediatory samples may well exceed the numerical limits of the up-sampling hardware and hence drive the interpolator into an overloaded state. How a DAC or SRC handles these intermediatory samples may well determine the overall sound quality of the device and why one device sounds better than another.
To this end we have designed a very unique system for catching and adjusting for inter-sample overs in real time so that only the necessary amount of attenuation of the incoming signal is used to limit the SRC from being over driven. We also provide a statistics mode which can be used to observed the amount of inter-sample overs on any recording. You will be surprised at how often and the magnitude of these overs occur in standard recordings when up-sampling is being used.
Of course all of this capability would be rendered useless if it could not be customized for the application at hand. To this end we have employed the expertise of DSP Concepts of California to incorporate Audio Weaver as part of our on-board firmware. Audio Weaver from DSP Concepts is an innovative design environment for developing optimized embedded audio software. It embodies years of audio product development experience and enables algorithm and product developers to more quickly and efficiently develop products and technology.
Typically to program a DSP would require a thorough understanding of the underlying hardware and digital signal processing concepts along with years of programming experience in a high level language such as ‘C’ or C++. In other words, not shy of having a degree in electronics or computer science for starters! Audioweaver takes this skillset and wraps a simple to use graphical interface around it so that people who have never programmed a DSP before can now do it in minutes! No cryptic ‘C’ compilers and linkers, just drag-drop and then upload to the board and you can build and customize your DSP designs like you had been doing it for decades! No need to hire a team of specialized DSP engineers – hire Audioweaver for your next project.
Unlike other DSP vendors boards built with the lowest cost in mind we didn’t skimp on quality when the benefits considerably outweighed the increase in cost which is why we specced in a DAC that cost at least fifty times more than the cheap one dollar DAC’s typically found in other DSP products ! We’ve spared no expense when it comes to component selection using the best quality passive components where appropriate. Premium audio grade capacitors such as Nichicon, Roderstein, Wima and NP0/COG ceramic capacitors in both the signal path and power supply make sure signal fidelity is preserved. Other DSP vendors will hope that you don’t notice the skimp on quality and lack of attention to detail in their products but we believe you deserve better.
For the tube aficionados out there we have incorporated a number of software plugins which behave like tubes without the physical tubes installed in the system. Typically, a DAC or preamp would use tubes to add some of that classic rich tube sound and texture to the mix. Instead we have accomplished this using the DSP in the form of tube emulation and soft clipping. Using a DSP to mimic the sound of tubes is nothing new. The pro-audio industry has been using this technique in its studio mixes to emulate the sound of the great classic tube recording equipment of the past. We’ve just taken this idea one step further and applied it to the reproduction chain!
For the first time we provide classic SET (single ended triode) sound with relatively inefficient speakers and solid state amplification all without the inefficiencies of a class ‘A’ single ended tube amp of the same power rating. Now there is no need to own a pair of exotic horn speakers just to enjoy that classic SET amplifier sound! As long as your solid state amp never clips you would never know you were running solid state amplification! We also provide various tube configurations with different distortion profiles and grid bias settings so that you have total control of the sound. Gain matching between different settings means that you can actually do objective comparisons without being fooled by changes in level.
Clock Source and Timing
- Femto-Clock specced low phase noise precision clock generator
ESSTech ES9038PRO 2nd generation Sabre Professional Audiophile DAC featuring:-
- Hyperstream II Technology
- 64-bit accumulator with 32-bit processing
- Patented Time Domain Jitter Eliminator
- -140 dB Noise
- -120 dB THD
- Precision 32-bit volume control on all channels in 0.5dB steps
- Precision Gain matching between chips
- Seven preset reconstruction filter types to suit different applications: –
- Fast roll-off, linear phase filter
- Slow roll-off, linear phase filter
- Fast roll-off, minimum phase filter
- Slow roll-off, minimum phase filter
- Apodizing, fast roll-off, linear phase filter
- Hybrid, fast roll-off, minimum phase filter
- Brick wall filter
- Master mode where the DAC determines the timing and the DPLL is not used
DAC Post Processing:-
- DAC-IV stages completely redesigned using high current differential, ultra wide 180MHz bandwidth amplifiers
- Purpose built optimal bias servo control for each DAC output to establish the optimum operating point for each DAC,
- Native onboard DSD Processing in real-time with no software pre-processing required!
- Supports DSD from DSD64(2.8MHz) up to DSD256 (11.2MHz)
- No DSP intervention
Advanced user interface console featuring: –
- Full colored QVGA (320×240 pixels) LCD with backlit display and crisp fast pixel rendered graphics,
- Stylish full colored bitmap graphics and true-type fonts for that professional appearance,
- Smooth rotary encoder control for master volume and quick menu changes,
- Large display of master volume,
- Seven dimmable status LEDs for real time status reporting namely:
- IR receiver activity status
- DAC Lock status
- DAC Mute status
- Plugin active status
- Line Lock status
- Sample Rate Converter status
- Full hierarchical intuitive menus and navigation system including: –
- Tactile buttons for menu navigation
- Remote control navigation of menus
- Full color display with context sensitive display
User Customization of DSP firmware using drag and drop DSP design software:-
- Design complex DSP systems in minutes
- No complex DSP coding necessary
- Dynamic instantiation so no additional compilers or linkers needed
- Precision low noise, high performance filters not limited to low sampling rates
- Field upgradable
DSP plugins for 2/8 channel preamp modes includes: –
- Soft-Tube emulates the rich sound of many different tube types and operating points
- Soft-Clipping emulates the clipping characteristics of a tube amp
- Distortion less precision 32-bit sinewave generator for test purposes
- Image focuser adjusts stereo image width
DSP capability –
- 4th Generation Analog Devices Fixed/Floating Point SHARC DSP with SIMD
- 400 MIPS, 800 MMACS and 2400 M-flops performance
- 192KHz DSP audio processing rate
- 32/40 bit floating point format
- 32 bit fixed point format with 80 bit accumulator
- IIR filter hardware accelerator
- FIR filter hardware accelerator
- 5 Mbits on-chip fast dual ported memory
- 256 Mbits 100MHz SDRAM
- 2 x 16 Mbits Flash RAM
- 64 Kbits EEPROM
- ESSTech ES9038PRO 32 bit Sabre 8 channel Reference DAC
- 49.152MHz Ultra low phase jitter clock source
- Texas Instruments PCM1863 24 bit ADC with PGA
- 49.152MHz Ultra low phase jitter clock source
Digital Audio Input Processing –
Digital Audio Output –
Audio Inputs –
- 2 x Balanced Stereo Input Channels
Audio Outputs –
- 8 Balanced outputs or 4 balanced stereo channels
- 8 Unbalanced Outputs for 4 unbalanced stereo channels
Digital Audio Processing Modes –
- 2 Channel-Preamplifier
- 8 Channel Preamplifier/Crossover
Bit-perfect master volume control
Bit-perfect level control on each individual channel
Master Reference Clock
- 49.152MHz Ultra low phase jitter clock and clock distribution network
Custom Logic Hardware
- Precision Reference Master Timing Generator
- DSD Processing Engine
- Signal Routing and MUX
- DSP Glue
- LCD Driver
- LCD Pixel Renderer
- Firmware upgradable via PC and USB cable
- APOS DSP operating system
- Audioweaver framework from DSP Concepts
- Home Theater Bypassing mode including multi channel active speaker setup
- Native DSD processing in the digital domain without conversion to analog first
- Board Dimensions:- 226mm x 125mm (Add an extra 10mm to both dimensions to allow for the heatsinks)
4 x S/PDIF Inputs:-
- 3 x Isolated COAX BNC Inputs (Zin=75 ohms)
- 1 x AES/EBU XLR Inputs (Zin=110 ohms)
- 192K sample rate capable
2 x S/PDIF Outputs:-
- 1 x isolated BNC COAX RCA output
- 1 x TOSLINK output
2 x TOSLINK Optical Inputs:-
- 96Khz capable
Dual function I2S HDMI input:-
- 192K sample rate capable
- PS Audio compatible
- Also functions as Perfect Link input to connect multiple preamps together
- Switchable isolator supply for fully isolated or partially isolated connection
Dual Function 1 x 12V Trigger Input/RC Input:-
- Isolated 3.5mm socket
1 x 12V Trigger Output:-
- 3.5mm socket
- 8 x Balanced Outputs or 4 stereo channel pairs
- 8 x Audiophile grade 3-pin male XLR connectors
- Conforms to AES-48 standard
- 100 ohm differential output impedance
- Maximum output level of 6V RMS + Gain Setting
- Independent channel gain settings from 0 to +9dB in 16 switchable steps
- Each channel is buffered by an additional high quality differential amplifier thus not inadvertently loading the preceding DAC IV stage
- Local capacitor multiplier regulator stage for pure DC supply
- Relay muting to eliminate pops and thumps from a cold start or mains switch off
- 8 Unbalanced RCA Outputs or 4 stereo channel pairs
- 8 x Audiophile grade gold plated RCA sockets
- 50 ohm output impedance
- 2V RMS max output level
- Independent DC/AC coupling selection for each channel
- Relay muting to eliminate pops and thumps from a cold start or mains switch off
- 1 x Balanced XLR Line inputs
- 1 x Unbalanced RCA Line inputs
- 1 x Unbalanced RCA Line/Phono MM Inputs
- 1 x MM Phono Input with selectable impedance and RIAA curve
- Switchable Moving Magnet cartridge loading options
- Resistance 33K,47K,68K
- Capacitance 50pF-800pF in 50pF increments
- Balanced Microphone input with switchable 48 Volt Phantom Supply
- 1 x MM Phono Input with selectable impedance and RIAA curve
- Linear power supply design with no switching regulators used for the analog stages
- 50VA low noise Premium Grade Toroidal Transformer
- 115V-230V AC internally switchable
- Mains DC blocker for optimum transformer operation
- Quality IEC connector with integral mains filter
- High speed Schottky rectifier diodes eliminates reverse recovery of normal silicon diodes
- Double pole RC filtering with over 10,000uF of total onboard filtering
- Four regulators providing +/- 15V, 48V Phantom supply, +12V Auxiliary supply
- 48V regulated Phantom supply for microphones
- Extensive chassis heatsinking for long life which diverts heat away from the capacitors thus extending their life
** Specifications and feature set have not been finalized and should be taken as a guide only.