SPL Channel One MK3 2130 | Channelstrip



Modification: Without Modification: Lieferung in 1-2 Werktagen

Modification: Lundahl Transformer: Lieferung in 3-5 Werktagen

  • Tube preamplifier with channel strip
  • Compressor, EQ, Transient Designer, De-Esser, Saturator
  • 2 microphone inputs with A/B switch
  • Preamp Out + 2x Outs for processed signal


For over 20 years, the Channel One has been synonymous with a high-quality and extremely musical recording and mixing channel strip.

In the latest Mk3 version, this classic has been fundamentally revised and, in addition to a higher internal audio voltage (now +/- 18 V) for an even more detailed sound, a further improved preamp section, an integrated transient designer, a tube saturation stage and a mic A/B comparison option for two microphones, it has many other great functions that raise the modern recording and mixing studio to a new level of quality. With the De-Esser, Compressor and Equaliser, all the important tools of a real channel strip are still on board. Whether microphone, line or instrument signal – the Channel One Mk3 makes every source sound like a professionally recorded signal.

The new design of the SPL Studio Series perfectly emphasizes the sonic capabilities of the third generation of the Channel One.

  • State of The Art Channelstrip
  • Mic preamp with 68dB gain
  • 2 microphone inputs with A/B switch
  • Lundahl Transformer (optional)
  • Preamp Output + 2 Outputs
  • 3-band EQ with AIR Band
  • Compressor
  • Transient Designer
  • De-esser
  • Tube-Saturator
  • Output-Trim
  • Frequency response: 10Hz – 200 kHz
  • Input noise floor: -126 dBu
  • Noise Mic In: -95 dBu at +30 dB gain


Transformers are transformers that are used as an alternative to electronic balancing stages in inputs and outputs. They are often used in sound engineering as they round off the fundamental range and add a little more pressure. Trebles sound more present and silky. However, in order to achieve maximum precision in signal transmission, electronic stages should be used. Therefore, taste and application are decisive here. If the Luhndahl transformers are used in the microphone input, they provide an additional amplification of 14dB and thus relieve the preamplifier. There is also no additional noise, as the ratio passively raises the level in the transformer. This is why the input transformer is more important for preamplifiers. However, the sonic development is only complete with both transformers together.

Compare two microphones?

No problem!

Channel One Mk3 offers two microphone inputs on the rear panel – Mic A and Mic B. Two microphones can be connected there. This makes work much easier. This means that you no longer have to unplug the microphone when comparing or changing microphones.

The 48 V switch activates the 48 V phantom power required for the use of condenser microphones. Phantom power can be activated individually for both microphone inputs.


The preamplification can be adjusted via the gain control. A pre-amplification of up to 68 dB can be realized for microphone signals – so even really demanding microphones can show their qualities.
The control range for line signals is between -20 dB and +16 dB. The control range for instrument signals is between -6 dB and +30 dB.

Preamp Out - directly to the destination

The Preamp Out picks up the signal directly behind the microphone amplifier. This signal can also be recorded on a separate track in the DAW, for example. If you realise after a recording session that the singer was a little louder in the perfect take and therefore the compressor was set a little too high, this incorrect setting can still be changed afterwards. The signal recorded for safety purposes can be played back into the Channel One Mk3 at a later time via the Line In and processed there with all modules.

Equalizer & Air Band

The LMF control is used to set the centre frequency of the semi-parametric bass/mid filter (LMF: Low/Mid Frequencies). The adjustable frequency range is between 30 and 700 Hz, so that this filter covers a range of around 4.5 octaves from the lowest bass range to the lower mid-range.

The semi-parametric MHF filter (MHF: Mid/High Frequencies) operates from 680 Hz to 15 kHz and also covers 4.5 octaves.

The gain of both filters regulates the signal by a maximum of +/- 12 dB. Both filters work according to the proportional-Q principle. This means that the bandwidth depends on the selected amplification or attenuation. This filter behaviour enables musically more meaningful processing of the frequency spectrum than with constant-Q filters: the more intensive the settings are selected, the more precisely the frequency range to be processed is determined in order to minimize influences on neighbouring ranges.

The filter labeled AIR in the EQ module is used to process the highest frequency range. A coil-capacitor filter with a bell characteristic is used here, with a centre frequency of 19 kHz. At this frequency, the maximum gain is +/- 10 dB. The soft and natural sound behaviour of the coil-capacitor filter is ideal for brightening up voices or instruments in the upper frequency range and thus improving their presence or giving them more “air”; conversely, sharp sounds can be given a more pleasant characteristic by attenuating them.

Technology of the SPL Channel One Compressor

In the Channel One Mk3 compressor, the parameters for the time constants (including attack and release) are set automatically and therefore adapt better to the changing conditions of the input signal than would be possible with manual control.

The attack time of a compressor should respond quickly with explosive P or T sounds and act more slowly with softly starting sounds, otherwise sound colouration will occur. The compressor in the Channel One Mk3 therefore adjusts more quickly for large level jumps than for small ones; long sustained tones are automatically processed with a long attack time.

The regulation of the release time is also dependent on the input signal. Fast and large level jumps are therefore processed with shorter time constants than small level jumps. Similar to the attack time control, the system adjusts with large time constants in the event of small level jumps in order to keep the distortion of the audio signal as low as possible. This technique represents the golden mean between fast, unobtrusive control behavior and minimal distortion, resulting in a natural and transparent sound. Another special circuitry feature contributes to the high sound quality of the compressor in the Channel One Mk3: the double VCA circuit. Two VCAs are used, one receives an in-phase signal, the other a phase-rotated signal. The two signals then pass through a differential amplifier. The effect of this circuit is to eliminate distortion products and offset jumps, as the interference can be eliminated by forming the difference between the two signals.

In addition, the individual VCAs share the workload and are therefore not even close to being overdriven – the risk of “offset” noises, audible as clicking or popping, is practically eliminated. However, the double VCA circuit also has considerably improved distortion and distortion values overall, so that a much clearer and more transparent sound is achieved than with conventional circuits. Voices and instruments sound much more natural and dynamic.

Transient Designer

With the Transient Designer it is possible to edit the envelopes of audio signals independently of the level (without threshold!). Boost or cut transients, lengthen or shorten decay times – with just two controls: Attack and Sustain! All time constants are automated in a musical way and optimize themselves adaptively according to the characteristics of the input signal.

The Attack control can be used to increase or decrease the transient phase of a signal by up to 15 dB. Positive attack increases the amplitude of the transient response; negative attack leads to a reduction in amplitude.

The Sustain control can be used to increase or decrease the decay phase of a signal by up to 24 dB. Positive sustain lengthens the decay; negative sustain shortens it.


SPL De-Essers use a unique technique to reduce S-sounds in a mix or on vocal tracks. In contrast to conventional de-essers, which use compressor techniques to influence the entire level, the SPL De-Esser works with filters that only process the sibilant frequencies to be reduced, but do not influence the rest of the spectrum. The sibilants in the unpleasant range are automatically recognized and mixed back into the original signal with phase inversion. In this way, an extinction is caused and the sibilant sound is reduced.
This method has considerable advantages, as it sounds very unobtrusive and largely preserves the original sound. Typical compressor side effects such as lisping or nasal sound do not occur.

The reduction is carried out by comparing the overall level with that of the individual S-sounds: the de-esser is only activated when the sibilant levels are above the average level of the entire frequency spectrum. Therefore, for example, voiced S sounds with a normal sibilant component are not processed, but only the overly loud, voiceless S sounds are reduced – the character of the voice remains unchanged.

Another special feature is the automatic setting of the threshold value (auto-threshold function), which makes processing independent of the input level. If the speaker or singer does not maintain a constant distance from the microphone, the processing is still kept evenly at the set S-Reduction value. Conventional methods are dependent on the input level and work more intensively the shorter the distance to the microphone.

The SPL De-Esser therefore does not require permanent monitoring and readjustment. In addition, the De-Esser can therefore always be used before the compressor, as a reverse order would bring no advantage – for this reason SPL were able to dispense with offering a circuit to swap the order.

The EQ pre TD button switches the order of the Equaliser and Transient Designer: When the button is pressed, the Equaliser is placed before the Transient Designer; when the button is not pressed, the order remains unchanged.

The Tube Post button changes the order of the Tube Saturation in the signal flow: When the button is pressed, the Tube Saturation stage is located after the Equaliser stage and before the output stage; when the button is not pressed, the Tube Saturation stage is located directly after the Preamp and before the De-Esser.

All at a glance

The lighted VU meter shows the input level, output level and gain reduction. The VU switch on the right under the VU meter can be used to select which level is displayed.

For better and more intuitive visualisation of the signal levels, the display range of the VU meter can be adapted to different signal levels. When the switch is set to 0, 0 dB on the VU meter corresponds to an output level of +6 dBu. In the +6 switch position, 0 dB on the VU meter corresponds to an output level of +12 dBu. In the +12 switch position, 0 dB on the VU meter corresponds to an output level of +18 dBu.

The OVL LED lights up as soon as an overload is detected in the device. The display interacts between all individual modules of the channel strip. This ensures that the sound processing only achieves positive effects and enhances every recording signal.

The DE-ESS LED lights up as soon as an S sound is recognised. It is only active when the de-esser is switched on, but works independently of the set reduction value. In other words, it always indicates that the circuit recognises sibilants.

Weight 7,6 kg
Dimensions 54 × 38 × 18 cm

Lundahl Transformer, Without Modification













Einheiten (HE)



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