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Hybrid
Includes both a Mac and Windows version of the program.
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VST-Plug-ins now enable you to use effect plug-ins as already integrated into Sequoia and Samplitude. Since their introduction these tools enjoy excellent reputation within the studio scene and are now also available for other VST-enabled audio products for the first time. Analogue Modelling Suite includes am-track analogue compressor, am-pulse transient designer and am-phibia pre-amp / channel strip.
The Analogue Modelling Suite provides you with first-class audio tools based on replicas of classic circuit designs. This includes a high quality analogue compressor (am-track) with optional simulation of a studio tape machine's saturation effect, a transient designer (am-pulse) for influencing the signal envelope of percussive audio material and a tube pre-amp/channel strip (am-phibia).
"am-track" is a combination of an analogue compressor and a tape simulator in one single device. It is primarily used for so-called "tracking", i.e. editing individual channel strips or subgroup signals. Compression can be executed either as modern or "vintage". With emulation of the tape machine sound, typical aspects of the large, saturated sound characteristics of the magnetic recording for the hard disk studio can be made reality at low costs and high quality.
"am-pulse" is a "transient modeler", a creative tool for editing envelope and sustain processes on percussive or dynamic signals. Furthermore, the signal can be subtly "analogued" or considerably distorted by modulating the subsequent tube saturation and treble exciter sections.
"am-phibia" is a tube amplifier/channel strip. It combines an optical compressor with a pre and post filter unit. As a result of the various different filter circuits, the device can be varied to a large extent and used as a vocal pre-amp, tube guitar amplifier or to create "warmer" sound characteristics according to the input signal and audio style. With this tool, the sky's the limit to your creativity.
The plug-ins of the Analogue Modelling Suite recognize the number of incoming signals and can function automatically in mono to reduce processor strain by preventing the sending of another signal for calculation.
am-track
am-track is a combination of an analogue compressor and a tape simulator in one single device. The plug-in was developed for targeted editing and enhancement of input signals (instruments, vocals). The audio signal can be given some more power and liveliness with a combination of compression and tape saturation in order to assert itself in the entire mix.
Compressor Section (am-track)
Two completely different compressors work in am-track, each with their own independent control and sound methods.
You may ask yourself why mention sound when talking about a compressor as the compressors merely relate to control actions. Unfortunately, it isn't as simple as the "making loud quiet" theory.
Various designs, algorithms and topologies for solving the actual problem (of the dynamic reduction) which all have their own unique character have come from the history of analogue and digital signal processing. For example, pre-filtering in the detector circle and the type of detection have a large influence on the audio results. Lots of hardware compressors have the same established VCAs (voltage controlled amplifier), but they all sound different and influence a signal, an entire production (or even a genre) with their "signature sound".
The two methods of the am-track can be selected using the switch 'vca / vintage'.
In VCA mode, the parameter selection and the circuitry's design correspond to a modern compressor with a VCA element as the control circuit and a forward automatic gain control in the detector section ('feed-forward design', that is, the controlling signal for level reduction is taken from the input signal).
The typical basic sound for this category is accurate, largely neutral and, in relation to the adjustable parameters, easily predictable.
In VCA mode the control signal is tapped at the input where it firstly executes a controllable low-cut (which can be set up via "detector hp freq" in Expert mode). The filter makes sure that deep-frequency signals have less influence on the adjustment settings; a popular trick for more power, for example, when using drums in a sub-group.
The filtered signal then arrives at the detector. With the forward gain control, previously set parameters apply fully and affect the adjustment settings immediately.
In contrast to this, there is a "feedback" method, which provides a second compressor mode.
Vintage mode
This mode appears as a preset on start-up of the am-track. It has less parameters than the VCA mode and has audibly less of a "surgical" quality, but more of a trimmed "sound character".
"Vintage" mode simulates a circuit design from the time where VCAs were not yet or could not be fully implemented. Instead a FET (field effect transistor) is often used as a variable resistor. This, together with a constant resistance at the circuit's gate, builds a so-called voltage divider, that is, it builds a resistance change at the FET (caused by a change in voltage at its gate), resulting in a damping of the input signal.
A very simple detector circuit is used to activate the FET; which obtains its signal from the output of the compressor (that is, behind the whole control circuit). For older designs, this feedback loop provides a stabilization of the work parameters and is one of the decisive factors for the often-quoted soft and musical compression of exponents of this design, like, for example, the Urei 1176 or 1178. The control circuit sees the layout of its previous work and oscillates to the signal.
The "disadvantage": the set time parameters for attack and release depend slightly on the input signal. With some applications it's actually advantageous with vocals, bass or even drums (for example, subgroup, ambience, mics). Here, you should rely completely on your ear.
Because of the feedback topology the maximum gain reduction is usually lower than, for example, VCA devices with forwards detection, usually 20dB. This way, there is almost always a level-matching amplifier in the feedback loop. The "drive" fader of the am-track regulates the so-called feedback amplification. This can be so high that the detector will get saturated by a loud input signal, resulting in signal peaks being swallowed up.
Simultaneously the setting becomes more intense, as quieter signal also start reaching the threshold. You can creatively implement this state according to the situation, bringing about complex compression of the signal, which doesn't sound much like dynamic compression due to the transients that slip through and release at high "drive" levels. The stated release of the signal, technically known as a ratio reduction, is also brought about by the centerpiece of the circuitry: the FET.
Level reduction works wholly as a function of its characteristic curve as a result of the non-linear behavior of this element. The FET virtually comprises part of the input resistance of the compressor circuit. As a result, the input/output response curve does not create a plateau when "drive" is high, which would be the case with a reference line with a high ratio or even limiting. A saturated FET can no longer complete the job it was marked out to do, i.e. to keep its output at low Ohm values. Once again, signal peaks will pass through the entire circuit undamaged, but the average level can be compressed strongly. From a technical point of view, the control process appears incomplete, but sounds pleasantly open and airy depending on its application.
The entire detection is dependent on the spectral balance in the virtual am-track circuit, the highs are automatically less strongly compressed, so that even extreme settings sound less flat and more lively.
It's the same story with deep basses. On closer listening you'll find that with strong compression, the signal still retains its power, which would otherwise get lost if the envelope were to follow shortly afterwards. "Vintage" mode has another feature to offer: At the output of the compressor in the signal flow an emulation of a transformer-coupled matching-level amplifier can be found. This contributes to some, if subtle, non-linear distortions at high levels, but is very much frequency-dependent.
Compression Parameters (am-track)
VCA-Modus
The regular set of parameters of dynamic compressors is available in this mode:
- Threshold: the threshold above which dynamic reduction starts working.
- Check the threshold display if necessary (thr): If the input signal reaches the set threshold, the blue dash will move around the arrow symbols. If this dash moves upwards, the threshold is below the average level and compression is active. Inversely, if the dash moves below marking, the input signal becomes too quiet to be able to reach the threshold - compression does not happen.
- Ratio: a ratio (1:n) which specifies by which factor the signal should be reduced once the threshold has been reached. For example, if the threshold is set to -20dB and the ratio to 1:4, an input signal of -10dB will only be amplified by 2.5dB (10dB : 4 = 2.5dB).
- Attack: the response time; that is, how long the arrangement takes to execute the required level reduction. Short attack times intercept level peaks, longer ones let them through unimpeded (compression only starts after this).
- Release: The time allocated to the circuit to reach the normal amplification factor.
Note on attack & release:
In general, short attacks are used for moderate compression and making the transient response softer; longer times can be provided to keep the 'bite' of a specific instrument at larger compression rates or to make the sound a bit snappier. With more difficult sources, like for example, a very dynamic vocal track (for example, a ballad) you can use a longer attack so that the arrangement runs more smoothly and quietly; the release time can be audibly trimmed to the pauses or the song speed.
Short release time may be used for modern, aggressive 'close up' vocals, for example, where breathing sounds can be an important stylistic device and the voice should sound very full and compact.
Knee: With this parameter you can specify the shape of the characteristic around the threshold. A 'hard knee' means that the transition of 1:1 amplification for level reduction occurs abruptly, a 'soft knee' on the other hand starts much lower than the threshold and moves the characteristic softly into the reduction. A 'hard' knee is well suited for effect-filled, audible compression like with individual drum tracks; you should, however, consider using a softer setting for complex and sensitive sources like guitars, pianos, or vocals. The more complex the signal, the easier it will be to notice a difference. For less sensitive sources, less importance can be attributed to this parameter. Note that for 'soft knee' settings the 'threshold' value has to be re-adjusted as the compression starts at a much lower level.
Vintage-Mode
In this mode, you can intuitively (by ear) use the dynamic editing features with just three knobs. Do whatever you want, but keep in mind: Less is oftentimes more...
- Drive: You can use the "drive" potentiometer to control the amplification factor in the feedback loop, that is, the signal strength which the detection circuit calculates. Furthermore, the internal 'ratio' changes within a limit, the more 'drive' there is, the higher the compression ratio.
- Attack and release: The same basic conditions as in VCA mode apply here. However, not only do you change the actual control response time after detection, but also the 'temporal window' in the detector. Additionally, the feedback arrangement method does bring about a certain amount of unpredictability. You should expect less control over the device in this mode, but more leniency on its part.
Compression expert settings
Of course, you can efficiently compress lots of data with am-track without having to press the "Expert" button or try out additional options. However, there are a few "handy" parameters behind the front panel. This applies equally to both compression modes.
- Look ahead: am-track is always ahead of the signal. Here you can specify how many milliseconds you want to allow for the "look ahead". The audio signal path is delayed according to the signal route so that the detection circuit is fed first with the input signal (so-called look ahead delay). You can now increase the attack time and still avoid fast peaks. The latency compensation in ensures that other tracks in the arrangement are adjusted and now time delay occurs. For percussive, cool sounds you can even set the delay to 0.
- Detector hp filter: This low-cut filter is positioned before the detection circuit of the two compressors. You can use it to specifically exclude basses and mids from these rules. Complex signals with bass and hi information such as a subgroup or complete mixdown tend to produce less "pumping" artifacts. This is because low frequency signals have the most power and would thus always trigger the regulation and modulate other frequency ranges in the volume.
- Auto makeup gain: Normally you have to continuously adjust the output amplification to generate a "compression" at the same maximum level. This is done with an activated Auto Makeup. The volume difference that can be expected from the set work parameters is determined and applied as an output factor after master regulation. If you prefer to adjust the "classic" level reduction and amplification manually, you can deactivate this function.
- Adaptive release: Admittedly it's even easier to adjust the "release" parameters by letting the device adjust the transient time. Adaptive release is "semi-automatic: you can roughly adjust the release time and am-track reduces it according to the current "signal power": from "slightly slower" to "considerably slower". In "vintage" mode this regulation method is particularly intense as it affects the feedback loop. For instance, if you are editing vocal tracks or dense, complex material, it can sound "calmer" or more "musical" if adaptive release is activated.
- Capacity: Adjusting the "capacity" controller sets "adaptive release". The greater the capacity, the more sluggish the release adjustment. You can therefore influence larger parts of the transient characteristics. For instance, if you want to use vocals that have been "moved forward", you should use a short release time (maybe 80-100 ms) and a greater value for semi-automatic (e.g. 80). Inversely you can reduce automatic feed by switching the relation (smaller capacity, generally greater release time).
- Comp mix: Parallel compression is a popular "studio trick", particularly with complex material. Adding the original signal retains the transients and spectral balance of the source. You can add compression by turning the mix controller. A "mixed" signal is particularly discreet, more transparent, and less "squishy" with vocals, whereby the compressed part usually has a higher level reduction than without adding the original.
Tape Section (am-track)
The tape simulation in am-track comes after the compression section and offers you the opportunity to give your recordings an "analogue touch" by reproducing typical aspects of a tape recording.
What also happens here is that the magnetic storage space of the tape becomes exhausted and the signal distorted when the recording level is increased. There are other factors as well however, like, for example, pre and de-emphasis. Since storing to tape happens depending on the frequency, pre-emphasis makes sure the dynamic area is used to its full extent, according to a normed characteristic curve (for example, NAB, EBU). This is because the signal cannot simply be transferred unfiltered at high levels onto the media via the recording head.
The pre-filter creates a characteristic harmonic spectrum for possibly intended overmodulation by the user, which then changes according to the level of saturation.
With the de-emphasis circuit the pre-filter can be undone by an inverse characteristics curve on playback. However, distortion caused by tape saturation might shift the spectral balance.
Depending on the enormous complexity of the process of magnetic recording, additional factors also influence the acoustic result, like, for example, pre-magnetization (bias). This relates to high-frequency voltage swing (usually in sine form at 150 - 200 kHz) which is then applied to the tape via the erase head before the same part passes by the recording head a few centimeters later.
This erasure current ensures the equal orientation of the magnetic particles, keeps the hysterisis loop intact, which is important for the magnetization process, and functions qualitatively. Low currents do have a brighter sound picture as a result, but magnetization is inefficient and the maximum recording level is therefore quite low. On the other hand, a bias current which is too high is associated with losses in highs, but does allow for higher levels with less distortion.
In addition, memory effects when recording tapes are responsible for a part of the characteristic sound as time-dependent factors play a role in the feeding of the tape reel along the heads, for example, mutual inductance and self-erasure.
Regarding the simulation of these processes, MAGIX has concentrated on some aspects of the "real world" and have created a virtual "machine" in am-track, which permits the following interventions by using the controllers on the interface:
- Level: Sets the input level. You determine when the "tape" is saturated and how great the effect of the coloring/soiling is. You simultaneously gain more "loudness".
- EQ low/hi: Adjust the frequency response (spectral balance controller). You can choose whether you would like to have an output signal of high bass level or whether it should have more highs. This alters the pre-emphasis at the "recording end" as well as playback equalization. In "Expert" mode you can vary the employed frequencies for lows and highs for distortion and equalization. However, please note that the frequency response of the simulation will not be neutral even if the "EQ Low/Hi" controller is dead center. There will always be some slight frequency-selective amplification, but that can be referred to as a machine's "sound".
- Bias: Moving the operation point (bias adjust). Increasing the bias will result in a higher "recording level", "magnetic flow" but tape saturation will occur sooner. You will also increase the above mentioned loss effects, and the result will be a (dynamic) reduction of the highs. Turning the bias controller to the left results in the opposite effect: the highs are not reduced, but the signal level stays lower.
- Tape mix (in the "expert" section): Anything that applies to a parallel compression can also be applied to the tape section. Non-edited transients in particular are responsible for perceiving "speed", "liveliness" and "freedom", but may be lost in the event of too much saturation/overdrive. Mixing in the original slightly, lets you regulate the band quite far while still retaining the so-called attributes.
Tips for using Tape Simulation:
The harmonics generated during saturation can quickly lead to a state of "acoustic fatigue" at the listener, particularly for high-frequency rich material and/or a frequency shift at the benefit of the highs. In a 1:1 direct comparison with the tape section deactivated, differences can be heard more easily. Subtle settings are generally sufficient for complex signals in order to get a slight "analogue touch".
Use the tape simulation as a "peak stop" if you use one of the two compressors to compress the signals: Transients that the compressor lets through (e.g. when using longer attack times) can be gently blocked by "taping" the signal afterwards.
am-pulse
am-pulse is a "transient modeller", a creative tool for editing envelope and sustain processes (attack and sustain) on percussive or dynamic signals.
In particular, the input of the signal is an important factor in the acoustic perception. The so-called transients are extremely short oscillations in the make-up of a sound which enable the human ear to precisely differentiate sounds from one another and localize where they come from.
Alongside transient modelling this effect also allows you to enhance the signal or even distort it. For this to work in the am-pulse, individual steps as "virtual hardware" have been set up and designed using elements of models and components "borrowed" from the analogue world in order to achieve smooth, organic operation with character.
Possible examples of use for am-pulse are:
- Drums (for example, Kick, Snare, Toms, Subgroup signal): Increasing or decreasing the attack leads to a harsher or softer sound respectively, increasing or decreasing the sustain changes the room information with overheads or ambience tracks.
- Acoustic or electric guitars and basses (accentuate or dampen the attack, e.g. for playing with a plectrum)
- Balance volume fluctuations and explosive sounds for sound recordings
- Sinking background noises
Transients Functioning (am-pulse)
At first glance it seems as if there is a dynamic compressor working "beneath the hood". However, this is not the case. In comparison to conventional compressors am-pulse is level-independent. If, for example, you reduce the attack rate of a snare drum by 6dB, this is independent of how loud it was played. However, you have to determine a threshold as if for a compressor. The signal is not altered below the threshold, but increases as the "gap" to the signal grows.
The mentioned possibilities for using am-pulse are based on the principle of "transient recognition".
am-pulse continuously analyses the input signal by means of so-called "Envelope Followers" that scan the time-based signal. Several Envelope Followers are used in the attack and sustain section. Both have different attack and decay times. Attack and sustain can be safely recognized by means of continuous comparison measurements.
Internally the am-pulse uses virtual VCAs ("voltage controlled amplifiers") that generate a control voltage from the resulting envelopes. You can use the fader to apply its value to the attack and decay rate. This lets you either "multiply" the input signal by the control voltage (increase controller) or "divide" the signal by the voltage (reduce controller).
Only dynamic signals are suitable for editing due to transient recognition. A more percussive input signal results in a cleaner and more predictable operation.
Transients Parameters (am-pulse)
There are three faders on the front plate for processing attack and sustain:
- level (left channel): The level is increased above the line; moving the fader in the opposite direction reduces the level. In the centre position the signal stays untouched.
- level (right channel): As above. Both channel faders are coupled together by default (as are the internal "detectors"). You can edit the channels separately by depressing the key with the padlock symbol. The control voltages for both channels are then determined separately.
- length: This is where you determine for how long the signal is scanned and "held" in the respective section. A lower value results only in a short amplification or reduction sounding quite electronically. Longer times mostly sound more homogeneous; however, the signal might even be affected too much in general (particularly during the attack phase). Depending on the complexity of the initial material, the "length" should be adjusted with more care.
Saturation and HF Details (am-pulse)
As a creative tool am-pulse offers an adjustable saturation level and a "high-frequency details" module.
Both sections are located within the circuit after transient processing. Modeled sounds can thereby be processed additionally. There are many ways to combine these three sections to create unique sounds. A few of these scenarios are below.
With hf details exciter circuitry can be used to add artificial harmonics to signals at the set frequency.
The saturation section works similar to a tube pre-amp. It shows the typical "tube-like" volume compression behavior and "drives" the signal into saturation at high levels. The "saturation" controller mainly determines the input gain of the tube stage. When turned fully to the left, no saturation occurs and this section remains inactive.
With the amplification of the signal, odd and even harmonics are created. The output signal first becomes louder and "richer". It becomes rougher and "dirtier" as the amplification increases, and finally sounds drastically distorted at an appropriately high input level.
The saturation of the signal in this case is frequency-selective: How much has to be pre and post filtered depends on the setting of the "saturation" fader. The more "saturation" the higher the filtering and sound modification. The internal degree of distortion is dependent on the setting of the level fader.
With the mix fader you can set how much of the processed signal is in the entire signal.
Creative tips for working with am-pulse
There are a few presets to am-pulse which you can select from the preset menu of the console. Some of these are ideal starting points for the following experiments:
- Attack & Sustain are often contrary processes. If, for example, you would like to add more "bite" to electronic drums by accentuating the attack, experiment with reducing the sustain level. Both sections then often require less "drive" and the sound generally has far calmer effects as if you had only used one of the two sections but at double level.
- To increase loudness, sounds with a lot of attack but a low average level can only be "tamed" by means of the saturation level. A few "saturation" dBs are usually sufficient.
- Try to simulate several amplifier types: Combine the transient and saturation sections with a slight reduction of attack (about 2-3dB) and a slight saturation (maybe 6dB). The results are soft signal edges, slightly increased loudness and a bit more "life" in the signal. You could maybe add 1-2dB "HF details". The result would then come very close to a tube amplifier.
- If you would prefer slightly more filtering from the saturation section (the above-mentioned "loudness") but less "crunch", move the "input" fader down slightly and increase the "saturation" controller correspondingly. Keep an eye on the Peak Meter here too. Subtle changes already have a great influence on the overall impulse behavior of the signal and slight colorations can even be seen in the A/B comparison.
- Set the attack and sustain of the left and right channels to opposite settings so that the signal "moves" in the stereo field according to its own transient data. This "autopanning" even works with mono sources and is ideal for creating new and unique loops.
- Drastic changes can be smoothed with the mix fader. Among others, entirely new sounds can also be created. The result usually sounds different to a simply softened setting of the fader at 100% mix. You can, for example, use sustain with acoustic drums to break out of the room, but keep the integrity of the recording with targetted mixing of the original.
am-phibia
am-phibia is a tube amplifier/channel strip. It combines an optical compressor with a pre and post filter unit. By selecting the filter presets the suitable setting can be chosen depending on the input signal. While interacting with the compressor section am-phibia can be used as a vocals pre-amp, tube guitar amp or simply for creating a warm sound. am-phibia can be used as:
Typical areas of use of am-phibia are:
- an optical compressor (the classic LDR-based light sources are modeled)
- a valve pre-amp
- a sound preprocessor with corresponding EQ and compressor settings
- valve guitar modeler with preamp, sound processor and case simulation
- a signal processor for warm sound synthesis
- Audio mangler
- Audio refiner
- Valve Exciter
Signal Flow and Steps (am-phibia)
The compressor of am-phibia is one of the most gentle compressors available on the market and is ideal for difficult signals such as vocals. The fundamental circuit is an emulation of a classic opto-coupling design with a so-called feedback circuit.
For this principle a light source (LED, light bulb or electro-luminescent panel) is fed with the output signal. The emitted light is beamed onto a light-dependent resistor and the changes to the resistance result in a change in the amplification factor of the entire circuit (controllable voltage divider).
The light source is fed from the output instead of the input due to historical, practical reasons: the circuit gently oscillates to the signal and is very stable without additional settings, and operates very musically so that, in hardware reality, such a device can be built with a minimum amount of required components (to the benefit of the sound quality). Due to the overall lag of the opto-coupling system, optical compressors have a relatively sluggish release reaction. This is accompanied by the so-called memory effect of the opto-coupler. A longer exposure to light also means longer return times. Due to the feedback control, the control times and compression ratio also highly depend on the control times, and the compression ratio strongly depends on the input signal.
You can specify whether you want to tap the feedback signal after the gain stage or the second filter. Together with the pre-filter stage you can create filter-dependent compression.
The compressor can be switched from compression to limiting.
3) Prefilter
There are different types of EQ. Each is treated as an independently designed circuit.
Active A/B
This is an emulation of an analogue filter module which operates with positive and negative feedback loops. Here, A and B differ from one another in the corner frequencies of the bands. B is optimized for use with speech or singing.
Passive A/B
This circuit corresponds to the classic Baxandall network and is comparable to stereo systems and some guitar amps, whereby type B is slightly rounder in the mid range than type A insofar as the basses and highs are further apart by spectrum, and the mid range is further accounted for.
Variant A is intended for general applications, filter B is optimized for vocals. The mid range, which is also available here, is not available in a Baxandall network and has been made possible in the existing circuitry as cascading low and high-pass sections, much like in common passive equalizers. The interesting thing about this setup is the effect the individual steps have on the entire phase response, which contributes a lot to the unique sound of this circuitry.
Guitar passive
Classic circuits such as on Marshall/Fender amps. The parameters are highly dependent on each other (more highs = less bass, mids are also influenced). Much like with general "passive" circuitry, here too you can find complex phase responses via the variable mixing in of individual branches of the filter network with the typical charm of such a classical design.
Guitar active
A convertible partner for typically American high gain sound. The parameters do not influence each other much unlike the passive variant, although a slight amount of internal feedback is applied to give the circuit a bit more power. Of course, this also influences the phase response and this way can similarly offer its own character.
Bass passive
An all-tube Peavey bass pre-amp, whose circuitry is similar to diverse Marshall/Fender designs, was the source of inspiration for this EQ (T. B. Raxx, similar to Alpha). However, the bass branch does not depend as strongly on the other settings; however, there is a high level of interactivity otherwise. If the mids are "weakened", the sound turns quite "hollow", ideal for slap basses.
Bass active
Similar to "Guitar active". However, the corner frequencies are designed for bass guitars.
4) Gain
The gain level as an emulation of class A tube circuitry is the heart of am-phibia. On the outside all you have to do is set the Gain level. "Magic" happens on the inside. For one this changes the frequency response, resulting in the right limit containing much fewer highs.
Some make-up gain is used to automatically compensate the quite high amplification factor so that typical valve-like distortions and harmonics occur at the upper end without becoming too loud. Normally, such high-gain circuit simulations have a weak spot at such a high degree of non-linearity: so-called aliasing. The harmonics created by distortions ideally exceed the audible range.
At a sample rate of e.g. 44 kHz, nasty artefacts occur due to signal mirroring of the Nyquist frequency (22.5 kHz) back into the audible range (without harmonic relation). This is particularly annoying with distorted guitar sounds, for instance, if you "bend" at the 24th fret when playing solo and aliasing becomes apparent by a bending in the opposite direction. For this reason MAGIX always leaves the internal sample rate of the gain stage at 176 to 192 kHz, in order to minimize artefacts.
5) Postfilter
This section only provides a low and high band equalizer; however, you can adjust the cutoff frequency point of both.
Active
This has the same parameters as the pre-filter stage, except that the mid band is missing.
Passive cut
Baxandall (cut) network as in the pre-filter. Also without mids.
Exciter
Amplifies the low and high frequency ranges by frequency-dependent saturation. The control elements mix the saturated part with the original ("mix in") as common with traditional exciters.
Due to the additional harmonics created during saturation the exciter circuit sounds very different to a conventional EQ. For instance, you can "refresh" the highs in a way impossible for conventional EQs, for instance, because important spectral information is missing. Despite the immense sound possibilities you should use the exciter carefully as the ear tends to tire out when listening to signals created this way and due to the high energy density.
Cab-Simulationen
With these sound control networks, am-phibia is leaving the "official" EQ settings. The loudspeaker simulations (cabinets) imitate the sound of classic case/speaker combinations as common with guitar and bass amplification.
In line with the "analogue by design theory" of am-phibia a high circuit activity. An armada of non-linear amplifier stages, filter circuits for frequency-response changes and complex phase-shifting operate in am-phibia.
The reason why MAGIX is using this process, is best explained by having a closer listen to the loudspeaker sounds. First of all there is the sound of the loudspeaker which sounds differently depending on the level and has a characteristic frequency response, but is also dependent on its environment.
This speaker emits direct sound at its front, while the back of the cone swings in the opposite direction so that the waves enter the cabinet phase-inverted.
More sound-altering characteristics occur here: The sound is directed at the enclosure, partially absorbed (e.g. by insulation material), partially reflected and mixed with the direct sound. Resonance effects, such as bass reflex tubes, static waves, return effects on the speaker and so on, are also apparent here. For the speaker models there are added properties of typical models, such as bass reflex tubes, to the signal transit time, and thus achieve a "typical" sound on the one hand, and a kind of life of its own on the other, which would not be achievable with impulse responses. The distortion created by the modelled loudspeakers at higher levels, is also an important aspect in regard to the whole process being highly interactive.
6) Volume
Volume makeup using a valve circuit that adds some atmosphere. This circuit ensures that the output signal does not exceed 0 dB.
Expert View (am-phibia)
Opening this view mode gives you even more opportunities for editing. Some decisive sound parameters of the device are accessible directly from here:
Opto Mem
This changes how the optical coupling device addresses the compressor and thus the related memory effect. You can use the "opto-mem" controller to work directly with the parameters for the system lag and influence the degree of program dependency.
Transients generally have less influence on controlling than longer signals. If set to minimum, the circuit "quickly" recovers. If set higher, the release time also increases according to the duration of a loud signal.
Clipping
This function controls the behavior of all tubes in am-phibia. Soft clipping (0) generates soft sound characteristics in overdrive. Hard clipping (100) creates a more global volume, although it may sound rough if handled incorrectly.
If this setting is used with guitars, the sound of the entire device may change drastically, particularly in combination with distortion sounds. As a guitar player you can make the settings best if you use the volume controller on your guitar to control the distortion. You will thereby most likely find the setting where am-phibia harmonizes best with the input signal.
With different sound sources, such experiments are less difficult, but are worth looking into.
Character
This control unit works in connection with clipping and the gain button. Sometimes, however, this effect is very subtle, meaning that an immediate effect is not audible with all effects.
If am-phibia is used as a guitar amp simulator, the controller can determine most of the sound. "Character" basically operates on the tube bias (control voltage for regulating the electron flux within a tube). This way you can observe a slight bass increase and more even harmonics.
A higher "character" value with loud and high bass signals can easily add "oomph", although there is a risk of the sound being rougher in lower ranges. Discreet use of moderate gain settings can sound similar to "transformer-coupled" devices.
Pre MF freq
This controller allows you to change the mid frequency of the pre-filter section. The available range is dependent on the underlying model.
Post LF / HF
With these controls you can change the corner frequencies of all controls of the post-filter section (basses and highs).
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