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Produs de TOP in Hi-End !
Phono Pre Amplifier with LCR RIAA equalizer The phono preamplifier is responsible for extracting the last bit of information from the cartridge by providing optimal loading while maintaining very high noise immunity. After vastly amplifying the input signal and applying the standard RIAA equalization it interfaces at line level with the rest of the system via balanced or unbalanced lines. The LCR method used to apply the RIAA correction is very rare due to complexity and cost. Orpheus Features:
Orpheus uses two tube gain stages with zero feedback and passive LCR equalizer with 2 inputs on RCA and one on XLR connectors. All inputs are balanced differential unless the grounding option is engaged on the RCA inputs. As we switch the ground plane when switching inputs there are no hum loops and induced ground noise
The input stage is a super low noise D3a German Post tube used as a triode. Loaded by a constant current cascode it provides the necessary gain and output impedance to feed the custom passive LCR RIAA correction following it. The switchable input transformer interfaces to MC cartridges and has selectable gain and loading impedance.
The equalization is passive and the next stage is another super low noise tube loaded by a transformer to bring the signal up to the necessary level and provide complete isolation of this sensitive circuit from the following components.
The filter is out of any gain loops and sits between the two gain stages protected from any unpredictable influences. The equalization is realized through the use of constant impedance LCR filters using coils wound with OFC wire and BeeWax impregnated paper capacitors.
This way that the input stage sees a constant load, making its behavior predictable at any signal level.
The power supply of Orpheus uses a choke input full wave tube rectifier arrangement. This eliminates switching noise and any other disturbance that comes before the choke. The other benefit of this topology is that the choke keeps the voltage on the capacitor bank constant. In our case, this allows for a capacitor bank made with vibration resistant military paper in oil capacitors to last a lifetime.
All this is governed by a microprocessor switching the appropriate combination of relays and monitoring the status and incoming command.
The chassis is milled in house from solid plates of different thickness aluminum that are interlocked together to form a ring like structure. We have the power supply mounted on the left sidewall of this structure and the signal transformers on the right side. Two semi-flexible bars on which the motherboard and its heat sink are attached, which then support this ring.
The top and bottom plates link all sides of the ring with the flexible bars and the feet made from carbon doped plastic on flexible screws.
There are no screws visible anywhere on the chassis. And the chassis feels like a solid block.
Technology Tube amplifier design has over 100 years of history. It started with the simplest possible amplifier which is shown below (fig A). This is a `single ended′ common cathode amplifier, referred to in the industry as Singe ended triode or SET. This type of gain stage is the basis of our amplifiers and is recognized for its superior sonic signature.
Why do we use triodes? Audiophiles have been led to believe through published measurements at maximum output power and uneducated reviewers that single-ended triode amplifiers produce vast amounts of harmonic distortion. As a matter of fact triode vacuum tubes are by far the most linear amplifying devices in existence today. They produce the least amount of distortion, and that distortion is predominately second harmonic, which is the least obtrusive type for the sound. By contrast, pentodes produce greater distortion, and the third harmonic tends to dominate. A transistor looks at best like a very bad pentode. Why the output stage is push-pull and what would an SE be? To state the obvious, a single-ended circuit must operate in Class A1 or A2. A push-pull amplifier may be Class A1, A2, AB1, AB2, B1, or B2. Class A indicates that each output tube handles the full cycle of the audio signal, while AB and B allow some of the devices to cut-off during a portion of the cycle.
Subscript 1 indicates that no grid current is drawn by the output tube, while subscript 2 indicates that the output stage enters the grid current region of operation. In the grid current region, the impedance presented to the driver stage is abruptly lower, and drive power is required, not just drive voltage.
The grid current region tends to be rather non-linear load for the driver and most designers will avoid it.
Single-ended and push-pull circuits may be built with triodes, beam power tubes, pentodes, or the latter two in ultra-linear mode. We use exclusively Class A push-pull circuits for our output stages, there is a natural cancellation of even-order harmonic distortion products in this topology. The cancellation is not complete, but it would be unusual to see large amounts of second harmonic distortion from a push-pull circuit. Note that a push-pull circuit has no significant ability to cancel odd-order distortion products. If low distortion performance is required, one must avoid the generation of odd-order harmonics in the first place. A good triode tube meets this requirement.
In single ended operation there is no mechanism to naturally cancel harmonic distortions and available power output is greatly limited. The full DC current for the output tube(s) flows through the transformer primary and strongly magnetizes the core of the transformer. Thus, much of the core's ability to couple the audio signal is used up by the non-audio DC current, and causes the core to saturate asymmetrically with audio signals Adding parallel output tubes for more power directly increases the DC magnetization current, thus exacerbates the distortion problem.
To deal with this an air gap may be introduced into the transformer core. In most cases also a greater amount of core material is used, which in turn makes the whole unit larger. By increasing the size of the coils we soon become limited by parasitic capacitance and leakage inductance affecting the bandwidth of the transformer. The final result is either a higher degree of distortion (all harmonics with the second dominating, increasing with decreasing frequency), a measurably peaked frequency response, or both. As observed the best sounding single ended designs rarely reach above 20W and is practically impossible to manufacture a transformer of utmost quality for more than 80W in single ended triode operation.
Since the distortion in the single-ended transformer is asymmetrical, a system based around this type of amplifier might be more sensitive to absolute polarity.
Same problems are true for the push-pull transformers but there the power limit for the same size transformer is 4 times higher. Assumptions lead to wrong conclusions. Traditional theory gives feedback high marks. Consider that when the error signal is fed back into a non-linear amplifier, it multiplies the distortion order. For example, if an amplifier naturally produces second harmonic, feedback will create a second harmonic of that second harmonic, which is the fourth harmonic. If the basic amplifier has second and third, the fed-back amplifier will contain second, fourth, sixth, and ninth. As is well known, the higher orders of distortion are far more objectionable to the ear than lower orders, and odd orders more offensive than even orders. Thus it may be possible to lower the level of distortion products and still have the distortion be more audible.? The application of negative voltage feedback also reduces an amplifier's measured output resistance, i.e., it raises the damping factor. Here again, the measurement fails to capture the essence of things. In the case of a feedback amplifier, better control of speaker motion is said to occur because the speaker's excess motion creates a voltage (the back e.m.f.) which enters the feedback loop via the amp's output terminals. The amplifier then acts in a manner opposite the error signal to correct for it. However, like many theories, this is an oversimplification and, in practice, the opposite result may be obtained. Quite often the motion of a speaker's voice coil former may not match the acoustical output due to cone break-up and the fact that the motion of the coil former is being sensed by the voice coil which is a reactive element with phase shifts and delays. On top of that the back e.m.f. passes through a cross-over network, which will again alter phase and frequency relations. By the time the error signal reaches the power amplifier it is arguably an erroneous error signal. As the power amplifier attempts to correct for this signal, it may actually do the exact wrong thing with respect to the speaker's acoustic output.
Small-Signal Distortion in Feedback Amplifiers for Audio Bring theory to practice − no feedback. It is logical that a signal passing trough and amplification stage will have some distortion added to it. (In our case this will be almost only second harmonic.) Passing trough the next stage we will add distortion to the distortion generating a minimum amount of fourth order distortion and so on. (Pretty much like the effect of feedback described above)
Consider now the following: the usual preamplifier (tube or solid state) has 3 stages (input buffer, gain stage, output buffer), then the simplest power amplifiers have 4 stages (input, phase splitter, driver, output buffer ). And all this is dependent of the signal amplitude via an extremely nonlinear function.
In order to minimize this effect the preceding stage of any amplification stage should have at least 2-3 times lower distortion than the latter. (Would that be possible when we have 7 stages in the signal path?) Minimizing the number of stages reduces drastically the order of distortion and its inter modulation products.
The sonic result is vastly improved transparency and speed.
Our products feature the minimum sensible number of stages implemented with the most linear devices possible operating in the closest to theoretically perfect operating conditions achievable.
Result? You be the judge! The power supply concept. Amplification stages are only half of the story. As seen on fig 1 the power supply is represented by a single capacitor. This assumption alone has ruined many beautiful designs.
In theory a large enough capacitor is as close as you get to the theoretically perfect power supply. Well in practice is not so! You have a rectifier and the mains supply connected to it. For every cycle of the mains you have two things happening. You charge the capacitor trough the diodes and the mains transformer for a limited amount of time by connecting it to the mains supply with all its noise and garbage and then you discharge it tough the amplifier and load until the next charge cycle. So your power supply is constantly varying its value and for a portion of the time is connected directly to the polluted mains line. (don′t forget the fact that this capacitor is part of the signal loop!!!)
And this seems to be acceptable for all electronic design gurus?!! Now analyze this.
Here is an example of a good amplifier design and it′s power supply. By looking at the whole picture you see a coil (choke) separates the reservoir capacitor from the signal capacitor, thus preventing any noise from the mains reaching the signal capacitor and as a side effect keeping a constant voltage across it.
It is very unfortunate that no currently available commercial products feature similar topology. On first glimpse it looks simple yet it does all that is required by the psu to approach the theoretically perfect with minimum component count.
Clean power, clean background and no IMD. A further improvement to the above example is the use of a choke loaded rectifier. This reduces the current fluctuations (pulses) needed to charge the reservoir capacitors and stops the diodes switching noise from entering the signal loop. Although high frequency interference noise is not audible its inter modulation products from the interaction with the audio signal are very noticeable. So avoid it at all costs.
Now leap forward a hundred years and implement this technology with the most advanced components and circuits possible and you are pretty close to perfection. The extra and often overlooked paths off the signal. In all directly heated tubes the signal current passes trough the cathode of the tube with is specific resistance generating AC voltage across it. Its amount is dependent on the tubes used and operating conditions but it is ALWAYS there.
In almost all implementations today there is a large size smoothing capacitor across it producing a practical short circuit for the AC signal. This is justified by the designers in order to have a stable DC and no hum in the filament, but it ruins the operation of the cathode in ac.
The simplest solution was implemented again many years ago, it is a simple choke in series with the filament. This is the simplest possible solution providing separation of the ac path and the dc path. Simple yet almost always overlooked. It is practically impossible to make a good enough inductor with reasonable size and cost to tackle the task, but a modern circuit called a gyrator can achieve the same effect. The many sensors in the amp. Now that we have designed a simple, yet sophisticated amplifier building it in practice to perform to design specifications presents a number of issues on its own. One of the most overlooked aspects of design is the mechanical construction and component vibration control.
As you probably know sound is recorder by sensing the sound pressure with microphones. There are quite a few types of microphones but they fall in two main categories: electromagnetic and electrostatic. Electromagnetic types work by sensing the variation of a magnetic field and logically the electrostatic work by sensing the variation of a static electric field.
Well guess what, every wire in magnetic field is an electromagnetic sensor. Does not matter what the cause of magnetic field variation is (the wire vibrates in a constant field or the field changes) you have a parasitic signal entering the signal path.
Each capacitor is in practice also a microphone superimposing the environmental vibrations on the voltage across it. This effect is only dependent on the mechanical construction of the device itself. (Defining how microphonic it is).
This applies to vacuum tubes as well as to solid-state devices like transistors.
The only way to reduce those effects beyond the threshold of audibility is by proper mechanical design of the equipment layout and chassis and the selection of components with correct construction and not only electrical parameters. Fighting noise and emi/rfi. another source of impurities that might enter the signal path comes from the air. Known as EMI/RFI (Electromagnetic interference and Radio Frequency interference). This is the effect of every piece of wire working as an antenna for airborne electromagnetic waves. Those get rectified and produce signals the interaction of which with the audio signal produces inter modulation products and are very audible. Some of you old enough that remember AM radio and it′s specific background noise, this is what you get in your amplifiers. Quite often the source of this interference is within the chassis of the device itself (bad PSU design, transformer and rectifiers) and more often than not enters through the power supply or other attached wires (noisy power line). (sound of cables ….) Balanced??!! Here comes a way of reducing induced noises in the interface between components.
Transformers stop many of the polluters from entering the signal path by the nature of their operation and construction.
A good description of how and why is available in the white papers at: www.jensentransformers.com Isolated cells. Using transformers to separate each gain stage of the system from the outside world is a good means of securing clean signal handling. Resulting in high s/n and much more expressive contrasts in the music nuances and timbre. This is also one of the reasons why our amps have clarity unheard off before. Power − how much is enough? This could be argued a lot. But here are our observations: At normal listening level (approaching and exceeding a bit the natural level of sounds) The peak SPL reaches 110-112db. So a high fidelity system has to achieve this in the listener′s room. On average the listening volume is 86-93db, the higher levels being peaks in the music program. Consider also that the average noise level in a listening room rarely goes below 24-26db SPL. Now put speaker efficiency on the equation and you will get something like that:
If your speakers are modern high performance super duper design with 82-87db/watt sensitivity look for a power amp capable of 400+Watts. (most probably class D types)
If you have been more sensible and opted for a speaker with 88-93db you don′t need more than 100w to enjoy the full scale of orchestral pieces and the selection of fine solid state and Tube amplifiers the market has to offer.
If you have your house built around a pair of full range horns with 108-112db/w you might consider a spud tube amplifier with 3-6W to rock the neighborhood. (this is a single tube amplifier)
Any other combination will not lead to the desired result. Handling digits. There are three different aspects to be discussed in DA conversion.
First is the Conversion process.
Second are the processes before conversion takes place (DSP)
Third is the timing of it all or clocks. There are multiple books on each of the above subjects, so don′t expect an in depth analysis but more a guide for what to look for in a DAC.
Multibit vs. Single bit (bitstream) converters. Google it. There are 1000s of pages, but it comes down to this: Multibit when correctly implemented is better, but far more difficult to make at high resolution. Hence cost is prohibitive. There is no single chip solution as of today! There were some 20bit converter chips (burr brown) in production and they have legendary status among users. Even today there are new designs coming out around the first Philips chips (20 year old) claiming audiophile performance and unbeatable sound. We have taken the road of multibit converters and have requested MSB to develop a version of their discrete DAC chips to meet our requirements. You can read about their principle of operation on the MSB site.
The hype about high sampling rates and the argument for non-oversampling dacs.
Well in order not to enter this argument we decided to support both.
We support high sampling rates up to 384kHz accommodating all existing formats, yet we are not bats − it is the inter modulation and filter characteristics that are audible not ultrasonic signals.
Resolution of our converter is up to 26bit fed with a 32bit word.
It means that we process and covert the incoming data without rounding and re dithering.
Again to overcome all arguments and to accommodate for the user preferences we support multiple options for up sampling re-clocking and digital filtering.
Any signal processing on a 24bit word of data will generate extra data. Almost all other dacs and processors will truncate the data back to 24bit. We let it flow at 32bits keeping all information as insignificant as it might be. Otherwise those errors accumulate and after 3-4 processes (re clocking, up sampling, dithering, energy filters) those become significant and are clearly audible.
Apart from the data fed to the converter the other major influence is the time at which the conversion happens. You can google jitter and will be swamped with lots of meaningless pages. In essence you are interested in the cleanliness of your clock and short-term stability, also known as phase noise.
You will see companies offering rubidium, atomic, gps locked, tourbillion and hourglass clocks out there but a properly designed and manufactured quartz oscillator is pretty much unbeatable for audio. Having said that please note that less than 0.1% of the ones we have seen are up to standard. The most interesting property for us is low phase noise at the very low frequencies and high s/n of the clock signal with no distortion. (at present in development).
All conventional DACs require buffers and filters. Those are always implemented with Opamps. (even discrete opamps) They isolate the dac from the output and sum the outputs of multiple dac chips, convert I to V and other tasks, but it does not matter haw good they are, they always severely limit the sonic performance of the dacs. Whatever you have done up to that point performs at the quality level of the opamp used. (isn′t this sad). Fortunately on multbit dacs with low impedance resistor ladder (it has to be low resistance to keep the noise down) you can take the output directly from the conversion resistors. Taking it directly we can sum the multiple dacs with a transformer with multiple primaries and by taking the output from the secondary of the transformer we have a complete solution.
Isolation of the output from the dac, stopping any high frequency content passing trough the band limited transformer avoiding IM distortion and summing out of phase removes all common mode noise and artifacts you can call it „the magic of coils. Why no body does that − and why they do only a partial job. Transformers don′t measure that well. They have limited bandwidth and rising distortion at high levels and low frequencies. All those shortcomings are true, but would seem to be substantially better performing in dynamic conditions than ANY op amp. The effect is jaw dropping not subtle. What you pay for and why it is a bargain. As explained above our products are very complex and use very special parts at substantial procurement cost. The price of our products reflects directly the cost of materials involved in its manufacturing and the time required to assemble them. All technology development is funded by other activities and the knowledge is reused in the development of other solutions and products.
As specific R&D is undertaken to further the performance of the existing products and to optimize manufacturing process, this should bring better products without raising cost.
|Specificatii||Concepts The human vision, hearing and other senses are not absolute measurement devices. They compare the difference between a reference and the tracked signals, and it's changes over time.
The hearing system registers sounds and requires some time to assess what they are. It then starts to track the changes giving less importance (losing interest) to long and sustained signals with no change.
It adapts to the surrounding (residual) background and concentrates on the constant changes in signal properties. Your vision is attracted by motion or flashing lights in this exact same way.
This phenomenon allows you to follow a conversation between two people in a noisy environment or listen to the voice of a singer undisturbed by a busy piece. In your office you realize how noisy it is only when the noise stops.
If the background is quiet the following sound appears louder and clearer. Anything different from the pitch-black background attracts your attention. Or if your hearing has adapted to a residual noise you start losing information (your brain does not pay attention to it). To make an analogy it is like increasing the contrast in a picture by darkening the background instead of increasing the brightness. The bright light will blind you and you will not see better, but removing the strenuous light makes your reference (black) stable and the picture becomes more vivid and sharp. It's like watching a movie in the cinema with the ambient lights on or off. Let us put this in the context we are interested in. Music is written with notes and pauses that represent sounds and SILENCE. The quality and quantity of silence in between notes is just as important as the pitch and timbre in the notes themselves.
It is rather complex why we have all those artefacts in our precious music through the reproduction chain but it is immediately apparent when they are reduced, and simply amazing when they are almost gone. Serenity and expressiveness come to mind when you experience it. It just sounds right. Inspiration The works of Japanese tube gurus are a source of inspiration for us. Shishido San's designs with transmission triodes manifested trough Wavac Audio and Kondo San's creations at Audio Note Japan are the best examples of the Japanese vacuum tube art currently in production and the different approaches taken to achieve the same goal. Their attention to detail and manufacturing quality inspired us to try and do better. The works and concepts of another Japanese designer came closest to the direction in which we were heading. Never implemented commercially, however, the designs of Sakuma San are a tough example to follow. Directly heated triodes with inductive loads seemed to deliver the effortlessness and tonal richness we are after.
Later we would realize that the root of it all was Western Electric and their creations. Their amplifiers and speakers are something like Adam and Eve for high-end audio.
Through those simple designs, music flowed seamlessly but with some shortcomings. Many tried to improve on those through the years, getting rid of most problems by unnecessary complication. The goal for improved music reproduction mutated into engineering competition in achieving meaningless target figures in areas alien to music and it's reproduction. It is a bit like specifying string weight and length in the piano. It is just irrelevant. Learning from legacy Looking at quality goods, wines, and watches we see the same techniques used years ago are still in use today. Yet the well made modern products are incomparably better.
We have all seen the glass piano and bodiless electric violin − again victims of technology perhaps.
We try to use as much as possible of the knowledge accumulated in the field and blend it with our own experience and cutting edge high-tech available today, so that we may achieve what previous attempts have failed.
There are superb designs out there by many reputable companies that reproduce sounds like us but few if any reproduce silence as good. Motivation Many years of exploration in the world of high quality audio passed in listening, recording and comparing. Vast resources were wasted on “superbrand” products before we realized we can't buy the product that satisfied all our needs and felt the need to develop our own audio components. We gained experience with all available technologies and amplification circuits making headlines through the years and gradually filtered them down to a list of worthy contenders.
Peoples prejudice is the fuel driving most manufacturers marketing departments. Capitalizing on the customers lack of knowledge by trowing at him terms and figures that are absolutely irrelevant to a units performance and claiming benefits unrelated to the above.
We found numerous high-end products which promised the world, but most of them seemed to tell a different story upon auditioning. Some exhibited very good qualities but severely lacked in others.
Some products carried true innovation but these inovations were usually limited to part of the whole making it no better than the average. Ingenious approaches coupled with off-the-shelf peripherals and bad manufacturing quality was the norm, as well as a lack of understanding of the whole system.
We listened - we researched − and this led us in a strenuous 2 year R&D project in our own lab evaluating amplification topologies and circuits. Numerous prototypes were built, studied, and some destroyed until a pattern and a theory emerged.
This was followed by yet another year of hands-on research and tests on the influence of various components and their qualities on sound, involving thousands of man-hours of auditioning and comparisons.
We spent a considerable amount of time on mechanical construction and manufacturing methodology (this being clearly visible in our components). Vibration control, electromagnetic and electrostatic shielding, contact potential and many others followed. What we did had to be consistent and reproducible.
We have not discovered a loophole in the laws of physics. What we found is that people quickly forget the achievements of times past, just to discover them again at a much later date. It is all there, you just have to put it together. We don't believe patented ways of heating water will make the difference. We look for the simplest meaningful legacy solution, find its flaws (usually wrong assumptions) and fix them if we can. It does not matter how much better a modern solution is if it is based on the same flawed assumptions and makes the same errors but better.
Taking advantage of available 21st century technologies and materials allowed us to achieve what early designers called fiction.
True to our beliefs we decided not to re-invent the wheel, but used the works and experience of the very best and implemented their knowledge in our product. It took a lot of time and tests to determine where the truth lied.. We do NOT build our own components but we buy them from the people that make them better than anybody we know (and we know a lot). And as for the rest, which we could not buy at the required quality level, we manufactured those components ourselves.
We are probably one of the very few companies that go to this extent. We don't use a supplier's line of components for convenience but we use only the best part available for the specific design. For example, Plitron in Canada makes the transformer and choke in the Spartacus power supply and Tamura in Japan makes the signal handling ones. In Dionysos the honor goes to Lundahl for power and Hashimoto for signal. The same is applicable for tubes and other components. We could write pages upon pages as to why each component in our products is there, and why it was chosen.
Actually, there is no trick, no magic. It is a complex function of knowledge, attention to detail, common sense and an open mind that gets you there.
I hope what we came up with pleases your senses and sets you on a quest to seek those qualities in all products.