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Unique Jitter elimination technology:
Jitter has always been regarded as
the root cause for sound quality of digital systems to be inferior
to that of high-end analog systems.
In some traditional DAC designs, the clock
processing method is to follow and to restore the clock of the
signal source, and some designs also include the capacity of
reducing jitter, but with limitations when the jitter level is
This common solution allows getting good
measurements in lab tests. But in actual use, most signal sources
have large jitter levels, so that the resulting sound quality will
not be satisfactory.
Some DACs worth only USD150 measure better
then than some well-selling DACs of foreign brands worth tens of
thousands. Who would think that this USD150 DAC is better? This
reminds me of when the China Insurance Research Institute tested a
direct collision between two brands of cars not long ago, and
netizens commented on the car that scored higher in the test: "It
has not failed the test, but it has not won the actual battle
The following three product charts are from
three DACs of different brands, guess which one is the most
expensive? (The answer is at the bottom of this page)
In order to completely
eliminate the influence of jitter in the incoming signal, the R7 HE
MK2 uses an asynchronous clocking process.
First it reads and stores
a considerable amount of signal source data to RAM, discard the
clock of the signal source, and directly use the high-performance
built-in Accusilicons to clock the output from the RAM. Since both
data and clock are handled by the FPGA, the impact of transmission
on jitter is neglectable. The jitter level is essentially the same
as what is specified for the TCXO clocks used by the R7 HE MK2.
The key to this processing method is to
ensure the integrity of data transmission, which is our technical
secret. Realize that as soon as a single data sample within a
million is lost in the transmission, a sharp and obvious cracking
sound, like static, or a scratch on a vinyl, will be heard during
Mr. He is passionate about music and, too him, listening sessions
are a source of inspiration for new design ideas. Our cie does not
only have the advantage of having full control over software and
hardware technology, we also possess a correct understanding of
music reproduction. Without having to rely on outsourcing for R&D,
we can at any moment listen to the gears during development,
allowing the pursuit and realization of the most realistic sound
Because the local clock and the signal source clock are completely
isolated and run independently, the traditional jitter measurement
methods can not be applied, instead a special data acquisition
circuit is required to evaluate its level. For human beings, the
best test method is always through listening tests. A jitter-free
sound is very realistic and natural, as clear as pure water, and
Fully discrete parts are used in a truly balanced current
4 sets of fully discrete and
independent DSD hardware decoders.
8 sets of fully discrete R-2R DA
modules to form a two-channel balanced push-pull decoder.
2 top-notch Accusilicon TCXOs with
frequencies of 90 and 98MHz provide synchronization for the whole
unit, and they are applied to the playback of all data rates without
32bit / 384K asynchronous Amanero USB
interface synchronized with the FPGA. USB and HDMI modules equiped
with galvanic isolators and fed by two groups of independent
isolated power supplies, to avoid the mutual electrical
contamination between USB and HDMI.
Using the latest discrete servo
stabilized power supply, whose performance is comparable to
batteries after more than a year of calibration and optimization for
each part of the circuit.
The digital circuit uses these new discrete servo stabilized power
supplies, while the clock and the analog circuits are still powered
by pure class-A power supplies to achieve the best sound, as
determined during listening tests.
The digital circuits of the whole DAC
are comprised of 1 FPGA and 7 CPLDs (both programmable devices),
which allow separating the different functional circuits and
The FPGA operates in the parallel
data processing mode even at the fastest data rate, and supports
firmware upgrades to improve sound quality when new and better
design ideas are eventually found and implemented.
The input signal sampling rate is
Support of external 10MHz clock input
(default is 50 ohms, can be customized to 75 ohms ).
All digital mode settings can
be changed on the front panel live.
A firmware upgrade port is
featured on the rear panel of the machine (no need to open to take
off the cover to upgrade the firmware).
Features of R-7HE MK2:
It has been 4
years since the R-7HE was launched in 2017. Thanks to the fully
programmable circuits of the R-7HE, we have continuously updated the
firmware to improve the owner experience.
We will continue to
update the current R-7HE, even if we are now launching a new
hardware design with the R-7HE MK2.
The R-7HE MK2 was
designed and tested at the beginning of 2020, which was followed by
a year of continuous audition, software and hardware fine-tuning.
Productions and sales were not scheduled until we were fully
satisfied with how realistic the sound was. The improvement is quite
obvious compared to the previous generation.
1. Using the latest designed
discrete servo stabilized power supply to feed the digital circuits,
the noise level is comparable to that of a battery, but without the
dry and thin sound characteristics of it, and thus the product's
operating temperature is significantly lower than that of the
previous model. The clock circuit and the analog circuit are still
powered by a pure class-A regulated power supply for the best sound
results (New upgrade)
2. By listening to a
turntable and studying its sound for a long time, we have
successfully integrated the analog vinyl sound characteristics into
this product, this being allowed optionally through the front panel.
3. Both the USB and HDMI
inputs are equipped with isolators, and two sets of linear power
supplies to separately supply power to the USB and HDMI modules
before the isolators to prevent interference from signal sources.
a, USB uses a two-way
transmission isolator, which not only transmits IIS signals to the
FPGA processor, but also receives the synchronous clock signal sent
by the FPGA processor. The USB interface itself is no longer
equipped with clocks. The synchronous clock is applied to make the
signal transmission more accurate and upgrade sound quality to an
excellent level. The sound quality is now in fact better then with
the previous generation R-7HE fed by the DI-20 (not quite as good as
with DI-20HE however).
b, the HDMI module is now equipped
with an independent isolator to improve the sound quality of the
4. Display of the input signal
sampling rate function.
5. The 10MHz clock input allows the
use of an external clock generator.
6. The FPGA operates using the
parallel data processing mode.
The IIS signal is serial data transmission. Each data bit requires
one clock cycle. One frame of left and right channel data requires
64 clock cycles, so stability over a 64 clock cycles is needed.
The parallel mode only needs one clock to transmit and process the
32-bit data of the left and right channels, which greatly improves
the processing speed and is less affected by the stability of the
IIS input data (USB and HDMI-IIS) is
reorganized into two sets of 32-bit parallel data as soon as it is
inputted. After the SPDIF signal is demodulated, it is
also sent to the next level of processing through two sets of 24-bit
parallel data. DSD data is also reorganized into two groups of
64-bit parallel data processing as soon as it is inputted.
According to listening comparisons, the parallel processing mode can
make the sound clearer and more neutral, with better dynamics and a
more analog flavor.
7. The clock management design of the
new architecture makes the clock work more stably, which brings
higher transparency and more details.
8. DSD uses the built-in asynchronous
clock to enhance timing, which obviously improves the clarity and
dynamics of reproduction.
9. SPDIF supports DOP playback.
Fully analog regenerative power
A regenerative power supply is like a
private generator built into the machine. It can filter out the
interference on the power line to the greatest extent, provide super
pure power to the audio components and restore a tuneful and
impactful bass and a highly analog signal to the audio system and
In the implementation of the
regenerative power supply, the 120V/220V AC power is converted to
DC through the regenerative power input transformer, which is
then supplied to the regenerative waveform generator and driver
through a class-A parallel stabilized power supply.
A set of balanced regenerative
waveform generators produce an ultra-low distortion 50Hz sine wave.
Following a balanced gain stage and a powerful driver stage, three
regenerative power output transformers output pure voltage, both
stable in terms of amplitude and frequency.
Two sets of class-A parallel
regulated power supplies provide independent power supply for the
analog output stages of the left and right channels and the digital
audio parts with its different processing functions. Both the
regenerative waveform generator and the driver are powered by a
class-A stabilized power supply, so that their work is not affected
by changes in the mains and interference.
The class-A shunt regulator has a
very high input impedance to block interference, preventing
disturbance on the mains power line from entering the audio circuit
and affecting the sound quality, and also avoiding the crosstalk
generated by the left and right signals through the power supply
thus improving separation. The low output impedance has excellent
linearity and speed, which also makes for an excellent regulated
power supply design. Audio-gd's arguments on the design of
regenerative power supply:
1. Through the internal design
analysis and actual test comparison of AP SYS-2722, the distortion
of the purely analog regenerative sine waveform generator is much
lower than that of digital synthesis generators.
2, The digital drive stage can
have higher power conversion efficiency, but the fully analog drive
stage has a more analog flavor.
3. The regenerated waveform
frequency of 400Hz has higher power conversion efficiency. However,
the regenerated waveform frequency of 50Hz is farther away from the
sensitive area of human hearing than 400Hz, and the sound quality is
thus more mellow and natural with 50Hz.
The advantages and disadvantages
of R-2R DAC:
1, R-2R will not convert the
clock signal to the output signal.
2, R-2R is not sensitive to
jitter but Delta-Sigma is quite sensitive.
3, The output signal level of
R-2R is more accurate than Delta-Sigma.
1, R2R's harmonic distortion
can be quite low but not as low as the ES9038 PRO (Delta-Sigma)
2, The accuracy and inherent
glitches of resistor ladders are not easy to deal with.
R-2R design popular in the market:
Whether it is DIY kits or
factory products, R-2R has become popular.
In the low-cost DIY kit market,
the usual design is based on the old MSB technology, but only retain
the signal conversion part and discard the exquisite design of the
This design uses data serial input to
a shift register IC to convert data to an analog signal. It is not
able to solve the technical problem of R-2R at all. The performance
of this design is completely dependent on the accuracy of the ladder
The factory products in
the high-end market use quite complex technology to solve the
problems of R-2R, in order to achieve high performance and sound
quality. Some manufacturers use the serial control mode of the shift
The design in the figure
below uses an FPGA to control the step resistance switching in
parallel. With the parallel control mode, the step resistance switch
of each bit is individually controlled, so it has ultra-high speed
(parallel mode only needs 1 clock cycle to output all bits, serial
mode requires at least 8 to 24 clock cycles) to send or update the
data, and can correct the data at any time to achieve an output
signal with low-distortion characteristics, and solve the problems
caused by resistance tolerances and switching glitches.
The accuracy of the step
Many people only care
about the accuracy of the step resistances because they think that
R-2R depends essentially on the accuracy of the resistance.
Nowadays, 24 bit is a standard, but
can the precision of manufactured resistors reach 24 bit? Even with
only 16 bits, the accuracy requirement is 1/66536, and so 0.1%
(1/1000) of accuracy is completely inadequate. In fact, 0.01%
(1/10000) would still fail to meet the 16-bit requirement, let alone
Therefore, lowering accuracy of
the resistance is not enough to solve the problem. If there were
0.00001% resistors available on the market, it would meet the
requirement of 24 bits, but the inherent resistance of the switching
devices would completely wipe out this super high accuracy
We need to solve the
problem technically, not just improve the accuracy of the
resistance. But we still use ultra-high precision resistors in our
Importunacy of the FPGA/CPLD:
On important thing to
underline about the FPGA/CPLDs is that they are programmable logic
array devices. Nowadays, FPGAs have been used in many high-end DAC
products, such as the popular ROCKNA WAVEDREAM DAC.
Since 2008, we have used
FPGA designs in DAC products. This machine is composed of one FPGA
and 7 CPLDs at the heart of digital processing.
The hardware layout
inside the FPGA can be designed and arranged through software,
therefore the hardware can be upgraded through software updates.
As a benefit, this
design has a high degree of flexibility. It allows improving sound
quality, adding functions, and keeping the product up-to-date, all
through software (firmware) updates.
Responsibilities hor the FPGA/CPLDs:
1. the FPGA implements a
high-performance SPDIF demodulator (instead of using low-performance
SPDIF demodulator chips such as DIR9001, WM8805, AK411X, etc...).
2. Combining clocking and FIFO
technology to output data, it can be accurately synchronized to the
clock, and thus reject jitter.
3. Built-in 2X, 4X and 8X
digital filters and different NOS modes allow users to choose the
tone that best suits their personal taste.
4. Simulate the tonality of
vinyl disc playback through a unique design.
discrete output stage:
The last stage in the
signal journey is the analog output stage, and the output stage has
a decisive influence on the sound quality of the DAC.
No matter how excellent
digital circuit design is, without an excellent analog output stage
design, the sound quality will become extremely ordinary.
The analog output stage
is directly connected behind the DA 7 module, using only
through-hole components (not SMD).
The high-speed ACSS
amplifier is used for signal amplification and processing. The ACSS
amplifier is designed without negative feedback and works in the
current signal mode. It does not have to repeatedly transform the
signal between current and voltage like other designs.
The output buffer stage
is a single-ended pure class-A FET design, and the two groups are
connected in parallel to achieve a lower output impedance. All in
all, the output stage is working in a pure class-A state, with no
negative feedback at all, so that it can reproduce pure and lifelike
The DAC has four
built-in operational amplifiers to serve in the DC servo function,
so that the DAC can work without coupling capacitors, avoiding noise
and coloration. In the whole signal channels following the DA
module, no switching elements are used to achieve the truest and
purest sound quality.
Powerful power supply design:
high-performance R-core transformers with a total power of 295 W and
more than 70,000 uF of audio-grade filtering capacitors to ensure
the purity of the power supply.
10 sets of newly
designed discrete servo stabilized power supplies and 7 sets of
ultra-high-speed linear stabilized power supplies provide separate
power to the digital circuit of this product.
stand-alone class-A parallel regulated power supply separately
supplies power to the regenerative power driver and each part of the
All power supplies
have ultra-high-speed and ultra-low noise characteristics.
The DAC uses 5mm thick
aluminum plates to separate the digital section, left and right
channel analog sections and transformer section to avoid mutual
interference between them.
Even the IEC socket,
inside AC power cable, have assemble in the copper box and copper
groove for avoid effect the signal parts.
The analog parts of
the left and right channels are symmetrically distributed on both
sides of the digital part, with the same signal path length and
distance, which makes the sound quality more even and accurate.
The DA 7 module is
installed between two aluminum plates to avoid interference from
other circuits. The complicated layout and installation procedure
have been chosen for the clearer and purer sound quality they bring,
and to make the sound field wider.
Product A ：USD150 DAC
Product B ：USD5000+ DAC American P brand DAC
Product C ： USD10000 British E brand RIAA front