Technology
The alpha compressor offers a broad variety of technical novelties and finesses which cannot be found in other compressors. Every little detail was challenged and improved to achieve optimum performance in a systematic process. All features have been developed with a fast workflow and reproducible results in mind.
But sophisticated technology must not be rocket science and efficient adjustments should not take hours. Therefore we understand even the most elaborate circuit as a means to an end: producing an outstanding quality of sound (although we have to admit that the technology itself is a great source of delight, too).
In the end it is detail which makes the crucial differences. The following part offers a closer look.
| Analog dynamic LED display >
Fast and exact, but without hectic rush. The perfect meter.
| Auto Fast mode >
Perfect attack and release even for complex material.
| Block diagram >
A concise illustration of the modules and their arrangement.
| Current feedback amplifiers >
Discrete amplifier stages for excellent dynamic projection.
| Discrete power supply >
Source of every good signal – electric current deluxe.
| Integrated M/S matrix >
The (actually not so) secret mastering weapon.
| Passive Current Attenuator (PCA) >
Evolutional concept for extreme musical level adjustment.
| Pure class-A design >
Optimum operation mode for full sound without nasty distortion.
| Soft Clip limiters >
Ideal tool for successful protection of AD converters.
| Technical data >
The measurements of the alpha compressor at a glance.
| Thermal control of critical components >
Guarantees constant conditions and reliable recall capability.
| Niveau Filter >
Offers direct influence on the overall sound character.
| VOVOX Sound Conductors >
Hot wire for perfect internal cabling.
Analog dynamic LED display
The gain reduction meter is a very important visual tool for evaluating the operation of the compressor supportingly. A lot of devices make use of sometimes more, sometimes less precise VU meters. But because of the inertia of the needle these meters are only usable with moderate time parameters. Especially when attack and release are set very fast the VU meter is much too slow to show real values. On the other hand this kind of display is fully analog and can therefore show all interim values stagelessly.
Another popular form of meter is the LED chain which can be produced fast and easily. Unfortunately it has a disadvantage, too: Because of the chosen driver units the change between two values always happens abruptly. A single LED in the chain can therefore only show an imprecise value in a defined interval. Hectic flicker indicates that the actual value must be somewhere in between.
elysia solves these problems by introducing the analog dynamic display that combines the benefits of both VU meters and LED chains. This meter is based on LEDs, too, but a special circuit design makes it possible to show interim values by modulating the brightness of the LEDs. This means a true analog way of showing the operation of the compressor: It is very fast, but with fluent transitions. The user gets an important tool for precise gain reduction monitoring – finally the relationship between acoustic and visual perception feels just right.
Auto Fast mode
The adjustable time parameters attack and release are crucial factors for the control mode of a compressor. Therefore choosing the right settings is very important, but depending on the dynamic characteristics it can be a difficult task – no matter if you are processing single channels or complete mixes.
Take the electric bass, for example: Its strings produce a significantly higher level for a short moment when they are hit than when they are sustaining. In this case it is not easy to find the right attack parameter. If a very short setting is chosen, the compressor is able to 'catch' the short peaks, but on the other hand the sustaining signal will also be processed which may result in audible distortion.
The special problem with bass is that in case that the B string is played it produces frequencies as low as 30 Hz, which is equivalent to a period of 33 ms between the waves. The result is that a fast attack setting will have its effect on every wave anew, producing unwanted artifacts.
Longer Attack settings reduce distortions significantly, but now the compressor reacts too slowly on fast impulses. The intended compression now turns into an expansion, because the fast and loud impulse is not processed while the level of the sustaining signal will be reduced. A slap-style bass displays this phenomenon very well.
This is where elysia’s Auto Fast function comes into play. If you set the attack to 80 ms and then engage the Auto Fast mode, the attack time will automatically be shortened on fast and loud signal impulses (e.g. string hits fingerboard when slapping). The compressor quickly reduces the signal and prevents the signal from 'slipping through'. Afterwards the attack time directly and automatically returns to its original setting. In Auto Fast mode the compressor can be very fast, but only when it is really needed!
The release characteristics behave in a very similar way. Again referring to our bass example, you often have to accept compromises when looking for the right release setting. If you set it too fast, you will create distortions, if you set it too slow, you will loose drive and loudness. Again, the Auto Fast mode is the solution. It will shorten the release parameter for a short moment and then reset it to the original value.
This function only influences the time parameters on short and loud impulses, in all other cases the original settings stay untouched. In both channels, attack and release each have an Auto Fast switch of their own, providing a great deal of flexibility in using it depending on the situation. Especially when processing/mastering complex and difficult material you have a very effective new tool at your disposal.
Block diagram
The block diagram shows the signal flow from the input through the particular modules to the output. M/S matrix, sidechain filters, audio filters, mix stages, transformers and the Soft Clip limiters can be individually activated via relays. The compression stages feature both feed forward and feedback modes.
Current feedback amplifiers
One of the greatest challenges in the development of analog audio equipment is the design of the amplifier stages. The easiest way would be the use of industrial op-amps that allow building efficient circuits at a comparatively low price. When premium signal quality is the goal, though, these kinds of op-amps reach their limits very soon. This is because of the low voltage and quiescent current, on the one hand, but the main reason is the limited sonic quality that just cannot meet the high demands e.g. in mastering applications.
If it means business there is no way to get around designing discrete stages out of single transistors, resistors, diodes and capacitors. Of course this costs more time and effort, but this is outweighed by the important benefit that every functional and sonic idea can be put into practice in every single detail, though. elysia even takes one step beyond and drives all amplifier stages in current feedback mode. What is the benefit from doing so?
Audio amplifiers based on classic circuit designs operate in voltage feedback mode. The elementary parts of these op-amps are differential amplifiers with two transistors that permanently compare the input with the output and correct the output signal if necessary. However, these circuits have a very small amount of delay often resulting in very high frequency oscillation. This effect is caused by changed phasing in the high frequency areas – the pristine negative feedback becomes a positive feedback then.
In the majority of cases small capacities are integrated into the amplifier stage in order to slow it down. The idea behind this is to reduce the amplification of high frequencies. Sometimes further external capacitors are added into the negative feedback path to achieve this. The problem is that these capacities have a negative effect on the sound quality of the signals that pass these stages.
The corollary was designing a circuit doing without capacitors that would affect the frequency spectrum. Intensive research finally led to a concept that has all the advantages but none of the handicaps of a classic op-amp: the current feedback amplifier. The current is directly modulated by the negative feedback and therefore avoids the time delay in the differential amplifier. The overall signal delay is clearly shorter and the phasing will stay intact even in high frequency areas.
Featuring an enormous slew rate and true reproduction of impulses, the amplifier can handle frequencies up to several MHz. The enjoyable results are clean high frequencies, a very good projection of details and open as well as natural dynamics. In addition, this amplifier can be tuned to work at a very low harmonic distortion level or also add a pleasant amount of K2 if wanted. Therefore some of our amplifier stages have experienced further fine tuning depending on their particular purpose. The resulting sonic quality is beyond any doubt and shows that classic analog design is still state-of-the-art for audio processing on the highest level.
Discrete power supply
The beginning of every good audio signal lays in the power supply. Its influences on the sound quality are much greater than it is often presumed, because it has an effect on the complete dynamic performance. A proper design using the right components is the key to a device sounding very dynamical and wide or rather docile and close.
All active stages are feed from the power supply, and therefore it has a great influence on many parts of the circuits. A loud and dynamic impulse results in all stages needing current very fast and at the same time. For this reason the power supply must provide enough capacity to deliver the right amount of current immediately. Having that in mind, elysia uses high grade electrolytic capacitors with a total value of 44000 uF and further 2000 uF on every PCB module. The special circuit design keeps the high charging current away from the signal ground and results in a better voltage supply.
Further on we placed additional MKP capacitors into many parts of the circuits in order to provide the current needed locally in a very fast manner. The voltage for the electronics is precisely kept at +/- 30 Volts. The voltage regulation is built out of discrete circuits, keeping the noise of the power supply so low that one can hardly measure it at all. At -114 dBu it is much lower than the background noise of the complete unit itself.
The measured hum is cancelled by 40 dB better than standard voltage regulators are capable of. This is an important condition for clean and clear sound results. If there were higher degrees of noise and hum in the voltage supply, the amplifier stages would have to compensate for that and could not fulfill their actual duty as well as intended.
The diode rectifiers have their influence on sound, too, because they have to recharge the electrolytic capacitors for a short moment and have to provide current for all circuits while doing so. Therefore elysia uses very fast and acoustically tested diodes for that purpose.
Also, we chose a toroidal transformer completely capsuled in MU metal. The production is laborious and therefore expensive, but it keeps the magnetic stray field on a negligible level. The transformer could therefore be placed inside the unit, keeping the connection to the power supply electronics as short as possible. An over-dimensioned measure of 140 VA guarantees that the transformer is always able to provide the electrolytic capacitors with much more current than they will actually need.
Integrated M/S matrix
M/S technology is commonly known as a variant of stereo microphoning. This technique uses a microphone with cardioid pattern for the middle signal (M) and another one with bi-directional pattern with an offset of 90° for the side signal (S). The main advantage of this technology is its mono compatibility. For this reason FM radio stations also use M/S technology for transmitting stereo signals. When the stereo format was introduced, the compatibility to mono receivers had to be preserved and therefore the side signal L-R was transmitted on a sideband.
To create M/S signals the left and right channel of the stereo sum are added for the middle (M), whereas the side (S) is created by subtracting right from left channel.
M = L+R
S = L-R
To decode an M/S signal into stereo again, M is added to S for the left channel and S is subtracted from M for the right channel.
L = M+S
R = M-S
The integration of an M/S encoder and decoder into a compressor generates new potentials that classic linked stereo compressors can hardly offer. One of the main advantages is the separate processing of middle and side signals, of course. With it, you can make the spatial middle region more compact without corrupting the original stereo spectrum.
Another advantage of this technology is the distinct reduction of problems that result from physical component tolerances. When developing analog audio equipment, a 100% equal design of two stereo channels can rarely be realized. In addition, even the best components available have tolerances that result in the real values of two channels slightly differing from each other even if the controller settings are exactly the same. This problem especially affects equalizers and compressors needing equal settings in order to achieve a clear spatial projection and a stable stereo middle.
If these methods of signal processing are integrated into a high grade M/S matrix the component tolerances will then have their effects on the M and S signals. But this does not have negative effects on the later decoded stereo image, because in M/S mode the mono shares of the left and right signal are both using the same channel for processing! There is no further need for a 100% exact match of both channels (which can’t be done anyway), and level variations between left and right channel belong to the past from now on. It is hard to get along without this feature once you have discovered what it can do for you.
Passive Current Attenuator (PCA)
One of the most important components in a compressor is the one responsible for controlling the signal level. This is the point in which the main technical concepts differ from each other. In contrast to equalizers where filter stages can be realized on the basis of 'natural' components like resistors, capacitors and coils, the development of a compressor asks for performing a number of 'tricks'.
One very popular concept is based on the field effect transistor (FET). In a certain range of its characteristic curve this component functions as a voltage controlled resistor that can be used for compressor circuits when driven properly. A real classic of the FET fraction is the UREI 1176 Limiter. But especially with discriminating applications like mastering in mind there are some drawbacks of this technology. Fist of all, the limited dynamic range causes increased distortion at high levels. The spectrum of this distortion is strongly dependent on the gate-source voltage and the resulting amount of gain reduction. Furthermore, the component tolerances regarding the characteristic curve of FETs are quite high and differ from one type to another. For this reason the sidechain of almost all FET compressors is driven by the output of the compressor (feedback mode) in order to achieve a useable characteristic curve. Because of these downsides the field effect transistor is not the right choice for premium compression quality.
Opto-compressors are another favored variant. In these devices light dependent resistors (LDRs) are responsible for changing the signal levels. The LDR is illuminated by lamps or LEDs and changes its resistance in correspondence with the intensity of the light source. This is an easy way to build an efficient compressor, but this type of construction also entails some limitations. One of the major problems is the time-based performance: Dependent on the specific components used the period between the change of light and the resulting change of resistance can be as long as 10 ms. Modern control voltage circuits allow faster values, but even these cannot be as fast as a FET or a VCA. Beyond that the achievable maximum gain reduction is limited by the minimum resistance value of the LDR. Some compressor models of this type are treasured just because of this kind of way-out behaviour full of character. Anyway, specific tasks like mastering for example need a larger spectrum of controllable time constants in order to get a steady grip on very fast impulses, too.
A third alternative is the use of voltage controlled amplifiers (VCAs). In most cases these are special integrated circuits (ICs) that vary their amount of amplification depending on the control voltage. The main benefit is a comparatively linear character curve. In addition, they can also be used as amplifier stages, giving them a very broad range of application (e.g. in noise gates). Even so VCAs are not often found in high end gear. One reason is the limited number of manufacturers and product models that can meet the sophisticated demands of the audio market. Furthermore, these components are integrated circuits that cannot be modified in terms of sound. Finally, the ICs contain active stages that have a fixed (and often too low) quiescent current rate that can result in unpleasant distortion on high input levels.
Because of the specified downsides elysia has developed a new gain reduction component: the fully discrete Passive Current Attenuator (PCA). This circuit transforms the incoming signal into a current and is then able to reduce this current controlled by voltage. The control voltage can be compared to that of a VCA and benefits from a calculable characteristic curve. The core consists of sixteen discrete transistors being kept at a defined temperature by an exclusive heating system that avoids unwanted fluctuations.
The aim was to create a musical gain reduction stage that could meet even the highest demands concerning audio quality. Countless experimental setups and auditions were needed to finally achieve this goal. Unlike industrial ICs the PCA works purely passive – it has no internal amplifiers and only reduces incoming signals. Not until the signal leaves the PCA, a downstream amplifier stage converts the current into voltage again, and the make up gain is added by setting the correspondent control. The characteristic curve of the control voltage was optimized for mastering applications resulting in precise but gentle gain reduction performance. If more keen and spirited kinds of sounds are on the agenda, the alternative feed forward mode offers them at the simple push of a button, too.
Further advantages of the PCA are its enormous bandwidth and its extreme fast control speed. Even huge level changes are done in just a few microseconds – an important premise for a fast and flexible compressor. The PCA plays an important role in the overall sonic character of the alpha compressor. It enriches the music with natural overtones by producing a healthy amount of K2 and K3. The sound characteristics become fuller and especially inspire digitally mixed productions with that certain something.
Pure Class-A Design
In a class-A amplifier the output transistors are always conductive. If the output stage is a push-pull amplifier design, two complementary transistors are used to each amplify the opposite half of the input signal. Both transistors have a quiescent current that keeps them conductive even when there is no input signal.
If this is not the case or if the quiescent current is too low, the amplifier will work in class-B mode causing the infamous crossover distortions. They occur because of a small glitch at the joins between the two halves of the signal at the zero point. These distortions are very unpleasant because they appear with every signal change and they are completely independent from the amplitude of the signal. In addition, they are very easy to detect for the ear because their spectrum holds lots of harmonics with constant amplitudes.
To ensure that an amplifier is operating in constant class-A mode, no matter what its actual input is, the quiescent current must not be set too low. One of the great advantages of discrete circuits is that the developer can decide what this value should be.
elysia sets the quiescent current at 14 mA which is an unusual high value for audio circuits. Doing so, the transistors will always be conductive, even when there are small resistor loads and high amounts of current. Crossover distortions cannot occur. For comparison: The total current consumption of a typical audio op-amp NE5534 is 4 mA and the quiescent current about 2 mA. The measurable crossover distortions of these op-amps are very small, too. However, this can only be realized by high amounts of feedback that need more signal stages, having negative effects on the fidelity of the signal in turn.
To guarantee pure class-A mode elysia utilizes power transistors normally used in power amps in the output stage. The sonic result is a clear increase of immediacy and dimension – all signals sound more present and massive. With a value of 2 watt each, the emitter resistors are also over-dimensioned in order to improve the projection of details even further.
Soft Clip limiters
One of the biggest problems of digital technology in audio applications is to set the right recording level of AD converters. The main challenge is that they do not dispose of a level reserve beyond the maximum of 0 dBFS that could catch short peaks. In the moment a signal is digitally overdriven it will be irrevocably damaged, because the original structure can hardly be reconstructed afterwards. Depending on the source signal, this type of peaks will produce very audible distortions because the signal is snapped off and does not correspond to the original source anymore. In addition, this kind of clipping produces lots of new harmonics that do not always fit into the desired musical context.
For this reasons it is obvious that AD converters should never be overloaded. Usually this is a task for limiters that reduce the input level from an adjustable threshold preventing the output level from exceeding the desired value. For successfully protecting AD converters from peaks, the limiter has to work very quickly in order not to let fast transients slip through. Thus, opto-limiters are not always the best choice: The characteristic curve may look just alright for limiting, but the photoelectric elements are too slow for catching very short impulses successfully.
A better choice in terms of speed would be a FET limiter, for example. But the need to work very fast remains and the FET limiter may cause sharp and audible edges in the signal. Moreover, the set time constants will have a noticeable influence, too: if the release time is too long, for example, the limiter will also reduce signals that would actually not have been too high in level and therefore degrade the original signal structure.
elysia’s Soft Clip limiter was developed to solve these specific problems. It is specialized on reliably catching short and transient-like signals. But the technical principle is different from that of classic 'brickwall' limiters that completely avoid further level increases beyond a certain threshold. It is better to compare it with an analog tape machine that drives loud impulses into saturation, and by doing so acts like a kind of 'natural' limiter. The Soft Clip limiter is built out of discrete transistors and similar to analog tape its characteristic saturation curve results in rounding peak signals rather than cutting them off. This makes it an effective tool especially when the source material contains variable peak values.
Technical Data
Frequency Response: | <10 Hz - 200 kHz (-0.5 dB)
|
THD+N @ +15 dBu, 20 Hz - 22 kHz: Stereo Mode (Direct) Stereo Mode (Compressed) M/S Mode (Direct) M/S Mode (Compressed)
|
0.0039 % 0.009 % 0.014 % 0.034 %
|
Noise Floor, 20 Hz - 20 kHz (A-weighted): Stereo Mode (Direct) Stereo Mode (Compressed) M/S Mode (Direct) M/S Mode (Compressed)
|
-95.8 dBu -89.3 dBu -95.6 dBu -92.3 dBu
|
Dynamic Range, 20 Hz - 22 kHz: Stereo Mode M/S Mode
|
122 dB 118 dB
|
Maximum Input Level: Stereo Mode M/S Mode
|
+28 dBu +23 dBu
|
Maximum Output Level: Stereo Mode M/S Mode
| +27 dBu +28 dBu |
Input Impedance:
| 10 kOhm
|
Output Impedance:
| 68 Ohm
|
Thermal control of critical components
Some delicate circuit components can be influenced by the surrounding temperature easily. The main reason for this circumstance is the discrete transistors that can react very sensitively to variations in temperature (that can – depending on the place of installation and operating time – happen by all means).
With the T16 Heater elysia presents a system that arranges constant conditions and reduces the thermal fluctuation to a minimum. This system was inspired by high-precision measuring instruments. It features up to 16 discrete transistors in a massive copper ring that is warmed up to a definite temperature.
A bordering ceramic cap the isolates the copper ring and keeps it from cooling down quickly and prevents exceeding heat emission into the housing at the same time. Once the system has reached its working temperature, it only needs little current to keep it at the same level. An electronic control circuit is responsible for a small variance of only a few degrees. The procedure is known from high end tube gear: the alpha compressor should be granted an adequate warm-up-time in order to experience it absolute top form.
Niveau Filter
This filter is a specialist in changing the overall sonic character of a track in fine nuances. It features two controllers per channel and is capable of flexibly producing convincing results in no time at all. Whenever a classic shelving filter would be too limited and a fully parametric filter would be overdone, elysia promotes the Niveau Filter.
Its main task is changing the proportions between high and low frequencies. The mode of operation is quite similar to a pair of scales: Dependent on the gain setting around a tunable center frequency the high frequencies are boosted up to +3 dB whereas the low frequencies are attenuated by -5 dB maximum. If the gain controller is turned into the other direction, the highs will be decreased and the lows boosted just the other way round.
The filter used for this application is an all pass variant with flat frequency response that changes its phasing correspondent to the setting of the frequency controller. When the signal that went through the all pass is afterwards mixed to the original again, signals that are in phase will be amplified and out-of-phase signals will be reduced.
This is how the special frequency response of the filter comes off. Because the filter signal is mixed to the original, the genuine structure and impulse-response can be contained to a large extent. None of the amplifiers will be shortened in their frequency response, which results in an open and dynamic sound. By simultaneously boosting and cutting the selected frequency-areas it is much easier to influence the character of a track ('bright' vs. 'dark') compared to using other types of equalizers.
VOVOX Sound Conductors
To achieve maximum sound quality with a high end product like the alpha compressor, it is very important to use a perfect internal audio wiring, too. If the wrong material is dpended on in this place, audible losses will definitely occur. After several experiments we tested the VOVOX sound conductors which we liked immediately. The difference is obvious and a noticable improvement; therefore all alpha compressors are equipped with this cable in series since Mai 2007.
VOVOX sound conductors transmit the whole frequency range with minimal losses. This means sound becomes brilliant and tangible. At the same time, the bass range becomes more powerful and precise. Due to their special design, these cables are able to transmit hard peaks very directly. Music becomes more brisk, dynamic and powerful.
During the record and the replay, VOVOX sound conductors enhance the spatial reproduction of music. Even in very complex situations, the sound remains transparent with clear contours. Often a significant enlargement of the stereo base is observed. As the sum of many different effects, there are also noticeable improvements to the ‘charisma’ of music: The clarity and immediacy of the sound signals offer an aura of calmness, solidity and sovereignty.
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