BLACK BY BEL CANTO

819 990,-
FRI FRAKT
 
Fjernlager

BLACK BY BEL CANTO

"Black by Bel Canto Design is a breakthrough product and, because of its sound quality , the most intriguing one I've reviewed—and by now, that's a lot of products." - Michael Fremer

The BLACK system architecture, consisting of the ASC2 System Controller and two MPS1 DAC/Amplifiers, incorporates numerous Bel Canto developed electronic and mechanical design solutions aimed at preserving the delicate audio signals. Digital processing and asynchronous interfaces driven from our Ultra Low Noise Master Clock circuitry ensures external noise from sources cannot corrupt, modify or color the original signal. Bel Canto’s ST fiber interconnect electrically isolates the ASC2 from the rest of the system, preventing random external noise from impacting sensitive analog stages in the MPS1 mono blocks. This eliminates ground loops and electrical noise pickup from environmental RF noise sources ranging from WiFi, Bluetooth, computers, cell phones or any other noise sources. Black is void of extraneous electrical noise; you only hear the music.

ASC2 - Asynchronous System Controller
The ASC2 is the second generation and heart of the BLACK system. The ASC2 is not a digital to analog converter; it is a Master Clock/Controller and the only one of its kind. It could be described as the digital preamplifier, with analog and digital inputs and digital output. The easy to read programmable display allows you to select the best view or turn it completely off.

The volume control and input selector is controlled from the multi-function rotating wheel on the top of the ASC2, you can also use the IR custom remote control, or your iPhone/iPad and the custom iOS app. The master control is always ready to adjust volume, press it once and rotate to enter input, or press and hold to enter the menu, where you can set the system up that best meets your needs.

The ASC2 is an all-digital asynchronous controller with galvanic isolated inputs that eliminate the potential for noise to corrupt the fine audio signal from digital, networked and analog sources. The ASC2’s 64 bit DSP processor core controls system function, 32-bit volume and custom FIR Filters retains the highest level of accuracy. And at the heart of the ASC2 is the custom ULN Master clock circuit, ultra-low phase noise clocks with independent power supplies measure below 40 Femtoseconds, keeping phase noise and jitter to extremely low levels, ensuring pure sonic performance.

The ASC2 handles all audio inputs, ranging from a networked audio UPnP Ethernet, USB2, a suite of SPDIF/AES digital inputs and an Analog line input that is digitized with a high performance 24/192 ADC 2cm from the input connector. All DSP based control and custom filter functions are contained within the ASC2, along with tone and level controls that dials in your system to your optimized sound. Just as a high performance analog preamplifier is critical to the performance of a traditional audio system the ASC2 is critical to the superior musical performance of the BLACK system making sure the signal is ready for digital optical transmission to the MPS1 monoblocks.

The ASC1 is still available as a System option upon request with its own unique feature set.

MPS1 - Mono PowerStream
The MPS1 is a unique monoblock design with Ultra low noise Clocking, Digital to Analog conversion, Bel Canto's class A amplifier stages, and custom 1200 Watt Power Buffer in an optimized environment.

The digital signal from the ASC2 is asynchronously re-timed before conversion in the advanced segment fully balanced mono DAC, that combines the best of Multi-Bit and delta sigma technologies measuring below 0.0005% distortion and 130dB of dynamic range. The high current output from the DAC is coupled directly into the I/V stage and within 3 cm into Bel Canto custom amplification.

The buffer stage is a discrete, low gain, low noise buffer that can drive 40A peaks into a 2-ohm load. It adds virtually no noise or distortion to the voltage amplifier output resulting in an extremely transparent audio power stage using only 4 high current output MOSFET power devices operating in an efficient self-oscillating analog switching architecture adding neither coloration nor distortion. The chassis design isolates these critical analog stages within a controlled, optimized environment, ensuring maximum performance under real-world system conditions.

The BLACK system achieves a greater level of resolution by creating a greater synergy between technologies eliminating potential signal loss, so you hear everything as it was originally recorded in incredible detail, speed and musicality.

The System Technology
Our new AMiP (Asynchronous Multi-Input Processor) Platform manages all the inputs of the ASC2. It is a powerful custom asynchronous core with 6 processor units that consist of: 2-32 bit XMOS general purpose processors, 2-32 bit ARM processors units as well as 2-dedicated audio processors. These together handle the ethernet interface that works with Roon Ready and DLNA servers, USB-A host streaming input, a USB2-24/384 input, legacy digital SPDIF/AES/TOS inputs and analog line level and phono inputs. The AMiP platform handles DSD and MQA processes and our custom Time Optimized Filter with integer up-sampling. There is a high-performance MM/MC programmable phono stage and 2-line level inputs that feed a high dynamic range 24/192 ADC into the processor core. Dual free running Ultra-Low Phase Noise Master Clocks direct all the digital processing and strip any incoming jitter from external digital input streams. There is a 32-bit ARM core processor that handles the user interface and coordination of the AMiP Platform.

The AMiP platform feeds our HDR-II DAC core running on a dedicated power supply through a custom digital interface. This is where volume control is performed using 32-bit math, the HDR-II core achieves 127dB of dynamic range, higher than any analog volume control could achieve for completely transparent volume control function. The HDR-II also has an asynchronous interface with a free running Ultra-Low Phase Noise Master Clock positioned adjacent to the Digital to Analog Converters. It includes a 64-bit dedicated audio Digital Signal Processor to provide completely transparent Bass Management, sophisticated Tilt and Bass EQ functions and allows us to fine tune things like the RIAA response in the phono section. There is a secondary DAC (112dB dynamic range) that drives the line output and provides headphone and subwoofer output capability with bass management features.

Bel Canto’s High Dynamic Resolution-II (HDR-II) core is a highly refined Digital to Analog Core rendering high levels of definition and musicality. It is the culmination of more than 20 years of audio DAC design evolution. With the HDR-II Core we have improved the dynamic range, analog output stage design and digital filters while reducing spurious signal generation and jitter related errors. Our new HDR-II Core has demonstrated that we can increase the resolution of musical detail, and improve listenability and perceived quality of the recorded music. This is in contrast where ‘transparent’ and ‘revealing’ often means ‘bright’. The HDR-II Core presents the original music with high levels of clarity and emotion.

Our HDR-II Core starts with a deliberate and careful choice of DAC technology. This choice defines the ultimate achievable performance and is defined by several key specifications. Our research has directed us to the best DAC technology, confirmed through extensive testing and listening. Bel Canto chooses the most refined CMOS digital to analog processor to provide superior dynamic range through extremely low noise, distortion and jitter sensitivity. The analog output section takes best advantage of the underlying CMOS technology to achieve the highest levels of analog purity and uncompromised performance. This Advanced Segment Technology DAC combines both multi-bit and multi-bit delta-sigma technology to deliver superb performance with a dynamic range of 132dB and transparent linear-phase digital filters. The CMOS analog conversion operates in a Constant Voltage, Class A Differential Current mode for minimal distortion and best analog performance.

Through detailed FFT analysis we characterize DAC performance over the complete dynamic range, insuring that the DAC analog outputs behave in a true analog fashion. As with the best analog circuitry our DAC technology exhibits diminishing levels of distortion and spurious signals at lower output levels were most of the audio signals live. Our HDR-II Core analog behavior is contrasted in the following figures with an increasingly popular DAC technology that looks good on paper but does not deliver true analog purity over the audio dynamic range.

It is critical that a DAC reproduce low level signals cleanly with no added distortion or spurious energy above the noise floor to color the sound. Note how clean the graph of our HDR-II Core is in Figure 1 with only the 1kHz, -40dB test signal appearing above the very low noise floor below -150dB. The extremely pure results in our Figure 1 HDR-II DAC results in a very pure lifelike sound quality. This was a major factor in our original choice of this DAC technology and our continued investment in this technology.

Digital Done Right
The HDR-II Core digital design starts with our Ultra-Low Phase Noise Master Clocks. These superior clocks are free running and fed from dedicated ultra-low noise analog power supply regulators. There are 2 stages of asynchronous interfacing to ensure that the final DAC output is jitter free and independent from any noise on the incoming digital signals. Our custom MQA based digital filters define an optimized impulse response from all sources, MQA, PCM, Analog and DSD. Figure 3 shows this impulse response with a 44.1kHz CD test signal. Our custom digital filters are designed to largely mimic the time domain response of an audio signal traveling through 10 meters of air. This superior time domain response insures the best sonic performance for all signals reproduced by our HDR-II Core DAC technology. MQA, High Bit Rate PCM, Analog and DSD all approach the impulse response resulting from travel through 10 meters of air, retaining the critical time relationships in the original music.

Pure Class A Differential Analog Output
A critical advantage of our HDR-II Core architecture is how the Class A Differential Current Mode output from the DAC operates in concert with the custom balanced analog section to best preserve the original dynamics of the recording. The large constant-bias differential currents impose pure Class A operation to every part of the analog output stages, every precision resistor, film capacitor and amplification device is biased with constant current and voltage. This constant bias keeps all components in thermal stasis and insures that all the original signal dynamics are retained.

Our custom discrete high current Class A amplifier is used in the critical current to voltage (I/V) and differential voltage amplifier stages. This is a fast, low noise amplifier design using a single-stage folded cascode architecture with over 500mA of peak current output capability. It maintains constant very low distortion levels through the audio band because of its high open-loop bandwidth. This proprietary Class A amplifier design brings a further element of dynamic resolution and contrast to the HDR-II Core.

The I/V and voltage amplifier stages include custom 0.1% Z-Foil resistors and selected film capacitors for the highest dynamic resolution possible. No detail is spared to produce the best audio experience, combining resolution with unforced musicality and expression.

Careful PCB layout represents a final degree of design refinement that is especially critical when the circuit is capable of 130dB of dynamic range or more.

Phono
Our dual stage phono architecture is used in the ASC2. It consists of an Ultra-Low noise input RIAA amplifier with precision filter components and fixed 40dB gain. This is followed by a digitally controlled analog variable gain stage to adapt the RIAA gain for a broad range of MM and MC cartridges. Input loading is controlled from the menu so that you can load your MC cartridge with the optimum impedance. This new Phono Architecture is optimized for accurate, low noise RIAA filter operation and compatibility with many cartridges.

Sub Output
Bass Management functions include independent selection of a 2nd order Butterworth Low Pass and High Pass filter set. The main speakers can be run full range or rolled off below frequencies from 40 to 120 Hz. The Subwoofer Low Pass can be set for full range or frequencies from 40 to 120 Hz. This approach drives maximum flexibility to optimize the integration of a subwoofer. There is also +/-6dB of gain trim range on the subwoofer output.

Line Level Inputs
There are 2-line level analog inputs that provide 110dB of dynamic range and can be used for legacy analog sources such as tape, tuner or a Home Theater bypass where the volume level would be set to a high setting (84.0) on the ASC2 and volume is controlled through the Home Theater processor. These inputs enhance the ability to integrate the Black System into many different systems with many different signal sources.

The System Features
The ASC2 Controller offers a suite of features for the Black System. Our precision filter and equalizer functions use the power of a custom 64bit DSP engine embedded into the internal signal path. Bel Canto’s approach removes the possibility of any degradation due to noise, distortion or response errors introduced by traditional analog filters. In other words, our approach permits subtle equalization and tonal control without compromising the sonic integrity of the original signal.

The Tilt function allows you to trim High frequency and simultaneously boost low frequency equally (or the opposite) to dial in the tonal balance you desire. This is done by holding the axis point of 775Hz and adjusting in a see saw fashion the upper and lower frequencies. This is ideal for eliminating the minor flaws in recordings or in sonically challenged spaces.

Gordon Holt is on record as saying: “it’s far better able than most to correct for many of the worst and most commonly-encountered sonic flaws in recordings.”

The bass EQ gives you great control over low frequency energy below 200 Hz helping you to achieve the tonal balance you desire in room and with speaker placement. It operates over a +/-3dB range with 0.6dB increments (–5, Off, +5) providing subtle optimization of bass energy.

The High Pass circuitry, inserts a high-pass filter into the main speaker output path optimizing the performance of your speakers with the sub output. You can also independently insert a low pass filter into the subwoofer output to help control sub levels. The filters are adjustable anywhere from 40Hz to 120Hz (in increments of 10Hz), depending on the low-frequency extension of your main speakers and the size of your subwoofer. The provision of these high-pass and low-pass filters is far-sighted, as it enables far superior integration of your main speakers with your subwoofer than would otherwise be possible.

The control menu also has a ‘maximum volume’ setting. This allows you to set the max volume at any level you prefer and ensures no one can inadvertently increase the volume using an App volume control or the remote and do damage to your speakers. The Max volume can be changed at any time and comes from the factory set at 85.

The Ethernet connection on the rear panel allows you to update your unit firmware to the current specification at any time the unit is connected to the internet. This can be checked through the menu by the user. If there is an update you will be given the choice to complete the upgrade ensuring that your unit is always up-to-date or decline the update and perform it at another time. If you choose to complete the update the unit will display the latest software Rev’s on the display and let you know when it is completed. The unit will reset itself (you do not have to power cycle the unit) and be ready to play after completion.

Tekniske data

ASC2 Specifications
 

Maximum Data Input Rates:
24bit Data to 192KS/s:
AES, SPDIF, TOSLINK
24bits to 192KS/s and DSD64:
10/100 Ethernet
24bits to 384KS/s and DSD64/128 (DoP):
USB2 Audio
Low Level Outputs:
Line Level Analog
4.5Vrms with Bass Management
ST Fiber Digital Outputs
2 - MPS1 Compatible (Stereo), 2 - Auxiliary
MM/MC Input
Input Sensitivity:
MM: 2.5mV to 5mV; MC: 0.25mV to 0.5mV
Input Load:
MM: 47K ohms; MC: 50, 100, 500, 1000
RIAA Accuracy:
+/-0.25dB, 50Hz - 15kHz
THD+N:
<0.01% 1kHz A-weighted
SNR:
>70dB A-weighted
Line Inputs:
Maximum Input:
2.2 Vrms RCA
Input Impedance:
10K ohms RCA
THD+N:
0.003%, 1KHz
Dynamic Range:
110dB, A-weighted 20Hz-20KHz
Performance:
Dynamic Range:
127dB A-weighted
THD+N:
<0.001% 1W, 1kHz, 4 ohms
IMD (CCIF):
<0.001% 1W, 18.5:19.5kHz 1:1, 4 ohms
General:
Power Usage On:
15W
Power Usage Off:
0.0W
Power Requirement:
100-120VAC, 220-240VAC 50/60 Hz
Dimensions:
19" W x 12" D x 3.8" H (483 mm x 305 mm x 97 mm)
Weight:
40lbs. (18.2 kg)
MPS1 Specifications
 

Output Section:
Maximum Power Output:
1200W-2ohm, 600W-4ohm, 300W-8ohm
Minimum Load:
2 ohms
Peak Output Current:
40 amperes
Frequency Response:
-3 dB 0Hz-50KHz, all loads
Output connections:
1-pair WBT Nextgen 5-way binding posts, ST Multimode Glass Fiber
Input Section:
Input Connections:
Stream ST Fiber
Dynamic Range Stream Input:
128dB
Balanced input impedance:
20Kohms
Performance:
THD+N:
<0.001% 1W, 1KHz, 4 ohms
IMD (CCIF):
<0.001%, 1W, 18.5:19.5KHz 1:1, 4 ohms
General:
Power Usage On:
37W
Standby Power Usage:
0.4W
Internally Set Operating Voltages:
100-120VAC, 220-240VAC 50/60 Hz
Dimensions:
19" W x 14" D x 3.8" H (483 mm x 356 mm x 97 mm)
Weight:
45lbs. (20.5 kg)
Features and specifications are subject to improvements and changes without prior notice.