Multifunctional multichannel compressor for audio signals

Abstract

The invention relates to a multifunctional multichannel compressor for audio signals and to novel multichannel compression methods. Disclosed is a multifunctional multichannel compressor comprising a plurality M of audio compression channels (AKK-m), wherein each audio compression channel (AKK-m) comprises a controllable output amplifier (compressor VCA-m), and a compression controller (KSE) having audio control channel outputs (StA-m) that are connected to the controllable output amplifiers (VCA-m), wherein each of the audio compression channels (AKK-m) is assigned a linking device for linking to at least one other of the audio compression channels (AKK-m), wherein a compression mode selector (KMW-m) is formed and is able to be used to define a compression mode, namely an automix mode or a group compression mode, for each of the audio compression channels (AKK-m) or group of linked audio compression channels (AKK-m), wherein the setpoint gains for the individual audio compression channels (AKK-m) are each dependent on a sum level ascertained in the side channel processing of the compression controller and on the respective channel level in automix mode, or on a common constant in group compression mode.

Description

[0001] Applicant: STAGETEC GmbH 12.09.2025 Our Ref: P18.918WO Multifunctional Multi-Channel Compressor for Audio Signals The invention relates to a multifunctional multi-channel compressor for audio signals and novel multi-channel compression methods. It is known to use mixing consoles in audio signal processing that have a multitude of inputs for audio signals. In a mixing process, a multitude of audio tracks are fed to a mixing console. The audio signals of the audio tracks undergo a multitude of different signal processing steps during the mixing process and are finally combined into a single- or multi-track output signal with a specific number of audio tracks, for example in a final format such as mono, stereo, or surround. In this general process, it is common practice, among other things, to insert a number of consecutive audio tracks, such as those of a miked drum kit, consisting of the audio tracks kick, snare, overheads, etc., into the mix.Toms and hi-hats are combined into a group signal, also called a bus signal, i.e., summed into a separate unit or "group." The various audio tracks are often named after the instruments whose captured audio signals they represent. In a mixing console, the components intended for processing an audio track or audio signal are also called a channel or channel strip. Accordingly, the various audio signals processed in a mixing console are also called channels, with the term "channel" sometimes also being used synonymously with the corresponding audio signal of the channel. The process of assigning an input source to a channel strip, or a channel strip to a summing bus or a direct output, is also called routing. Multiple channel strips or direct sources can be routed to a common bus.by summing their signals with weights. Such a combined bus signal or group signal can then be subjected as a whole to subsequent steps, such as joint compression. Compression in the context of audio signal processing serves the purpose of limiting the dynamic range of an audio signal. This is achieved through controlled attenuation of the audio signal. Such compression of a group signal, such as that of the drum track, leads to a limitation of the dynamic range of the drum signal. Applicant: STAGETEC GmbH 12.09.2025 Our reference: P18.918W The group signal often undergoes a subjectively perceived compression or narrowing.which is a desired effect. An adjustable gain control can optionally compensate for the level loss caused by compression. However, the following problems arise: Often, one wants to add a separate reverb component to individual audio signals, such as a snare drum, and uses a send output for this purpose. A send output is a weighted tap of an audio signal. If the output signal of the reverb unit is mixed into the bus after passing through a group or master bus compressor, the ratio between the dry and reverberated signal varies, since the compressor modulates the volume of the dry signal, which is included in the master bus. This case is illustrated in Fig. 1 for three channels that send their signals proportionally to the effects unit. The reverb signal, or more generally, the effect signal, is referred to as the wet signal, and the dry signal as the dry signal. Therefore, a varying wet / dry ratio results.which, however, is often undesirable. One measure against this effect is to include the reverb signal in the sum of the group before the compressor. This is illustrated by way of example in Fig. 2. This approach has the disadvantage that the effects unit would not be used for other sources outside the group and that the effect signal itself is compressed, which is not necessarily desirable if a particularly authentic spatial impression is to be created. The invention is based on the objective of creating a compression device that expands the audio signal processing possibilities, in particular improving and / or expanding single-track effect signal mixing possibilities. The objective is achieved by a multi-channel compressor with the features of claim 1 and a method with the features of claim 10. Advantageous embodiments result from the dependent claims. The invention is based on the idea thatto create a multi-channel compressor in which each channel has its own compressor, i.e., a controllable output amplifier, as a rule, designed as a voltage-controlled amplifier (VCA). A compression control unit is designed to determine the individual gains or control voltages in a so-called "side-chain" processing based on the audio signals present in the individual channels. For this purpose, it is provided that one or more sum signals for one or more groups of interconnected audio signals or channels are determined in the compression control unit, which are subjected to common, interrelated compression, i.e., a dynamically limiting output gain. If such a sum level generated in the side-chain processing, that is, a determined level of this sum signal,If a threshold value, definable for the respective group compression, is reached, the output gain of all output amplifiers of the channels coupled into a group is reduced. To achieve this, individual channels can be linked together to form one or more groups, with each channel only being able to be included in one group. Such a compressor can emulate a bus compressor, but the individual audio signals / channel signals, which are later combined into a bus signal after individual compression, are each compressed to the same degree. However, they can still be used individually as a starting point for send signals, which can then be fed to effects units, for example, to generate effect signals for further mixing. Due to the fact that each channel has its own controllable output amplifier,A so-called automixer can also be implemented with the same multi-channel compressor without much effort. For this, it is only necessary that the corresponding channel levels are determined for each individual channel grouped together in the "side-chain" processing of the compression control unit. Instead of the constant threshold value used for all channels to emulate a bus compressor, the control signal for each controllable output amplifier is determined individually by comparing the sum level with the corresponding channel level. If the sum level in a channel exceeds the corresponding channel level,This reduces the output gain in the corresponding channel. In an automixer, the individual audio signals are thus adjusted and compressed differently with regard to their dynamics. Applicant: STAGETEC GmbH 12.09.2025 Our reference: P18.918WO Automixers are generally well-known. Put simply, an automixer's task is to distribute the gains of competing signals, such as those assigned to the speakers in a talk show, using normalization so that the overall gain of all signals, or the overall volume, remains constant and the loudest signal / channel receives the largest share of the gain. Passive microphone signals are generally suppressed to the extent that a relatively loud microphone signal from an active speaker is present.that the proportion of ambient noise captured by these open microphones decreases noticeably. Furthermore, the automixer increases the maximum gain of a connected loudspeaker system that is possible before feedback occurs. It is explicitly pointed out here that no bus signal is generated in the multi-channel compressor. Only in the side-channel processing, i.e., in the compression control unit, is a summed signal generated from the audio signals that can later be combined into a bus signal, and its sum level is determined. However, this serves only for control purposes and does not represent direct signal processing of an audio signal in the sense of a bus signal. This summed signal is also referred to here as a virtual sum or virtual sum signal. However, this is a real, generated signal.However, this is only a control signal, or rather, it is only used for control signal processing and not the "actual" processed audio signal. The resulting multi-channel compressor is thus capable of executing different compression methods, even simultaneously, for different channel groups. Since the channel mapping is freely configurable, the number of group compressors and / or automixers depends solely on the group size(s) assigned to the individual compressors or automixers, as each channel can only be assigned to one group. If a channel is not linked to any other channel, it can be compressed individually according to a threshold that can be defined for the channel, or preferably, compression can be deactivated.so that the output gain is performed independently of the level of the respective channel. Applicant: STAGETEC GmbH 12.09.2025 Our reference: P18.918WO In particular, a multifunctional multi-channel compressor is thus created, comprising a plurality M of audio compression channels (AKK-m), wherein each audio compression channel (AKK-m) includes an input (Em) and an output (Am) and a controllable output amplifier (compressor VCA-m) connected between them in terms of audio signal technology, and a compression control device (KSE) which has for each of the plurality M of audio compression channels (AKK-m) an audio control channel (ASK-m) whose audio control channel input is connected to the input of the audio compression channel (AKK-m), and an audio control channel output (StA-m) which is connected to a control input (StE) of the controllable output amplifier (VCA-m),wherein each of the audio compression channels (AKK-m) is assigned a linking device for linking to at least one other audio compression channel (AKK-m), wherein a compression mode selection device (KMW-m) is configured to define a compression mode for each unlinked audio compression channel (AKK-m) or each group of linked audio compression channels (AKK-m), wherein the compression control device (KSE) additionally includes at least one summing device (SE) for forming audio control channel summation signals (sp(n)) (AVS-p) from interconnected audio control channel signals (cm(n)) of the majority of the audio control channels (AKK-m) and at least one detector device for determining channel levels (D, m (n)) for the individual audio control channel signals and sum levels (S p(n)) for the audio control channel sum signals (sp(n)) includes, wherein the compression control device (CCD) has a channel gain calculation device (CCD-m) which provides for each audio compression channel a target gain signal (Hm(n)) as a function of the sum level S p (n), into which the audio control channel signal has been received, and a comparison threshold is calculated, wherein the comparison threshold is selected depending on the compression mode set by means of the compression mode selection device (KMW-m) for the corresponding audio compression channel (AKK-m), wherein in automix mode the comparison threshold is the channel level (Dm(n)) and in group compression mode a fixed threshold constant (C) p) is. Furthermore, a multi-channel compression method for a plurality M of audio signals is created, comprising the following steps: Applicant: STAGETEC GmbH 12.09.2025 Our Ref: P18.918WO Feeding audio signals into a plurality M of audio compression channels, each of which has an input and an output as well as a controllable output amplifier arranged between them; Establishing connections between the individual audio compression channels to form groups of channels; Determining the individual gains for the output amplifiers of the individual audio compression channels by forming a summed signal from the audio signals of audio compression channels grouped together in a side-channel processing (side-channel processing) and determining a summed level based on the summed signal.and the control signals representing the gains for the individual audio compression channels are determined by comparing the sum level into which its audio signal has been incorporated with a threshold value, and wherein an attenuation of the gain occurs when the sum level exceeds the threshold value, wherein for each channel group a compression mode is selected from a set comprising a group compression and an automix, wherein, when the automix compression mode is selected for a group of linked audio compression channels, a channel level is determined for each of their audio signals, and the channel level is used as the threshold value in determining the audio compression channel-specific gain, and a compression constant is set for a group and is used as the threshold value in determining the audio compression channel-specific gain.when the compression mode for the group is set to group compression. A preferred embodiment provides that each of the audio control channels includes a side-chain filter (SFC). These optional filters expand the possibilities for sound shaping. Referring to the example above with a multitude of drum audio signals, the influence of the bass drum audio signal can be reduced by attenuating the bass frequencies using such a filter. These filters affect the achievable sound. Another possibility for sound shaping, particularly through specific instrument tracks, i.e., specific audio signals, is created in one embodiment.where each of the audio control channels includes a weighting device (Wm). Applicant: STAGETEC GmbH 12.09.2025 Our reference: P18.918WO Such a weighting device assigns a weight to the audio signal for the corresponding audio control channel when calculating the sum signal and thus the sum level, and also influences any channel level determination that may be performed. Adding weighting allows the influence of individual signals on the group's compression to be increased or decreased. An example would be a relatively loud bass drum, whose influence would be reduced by weighting it below 0 dB. Since each signal can be weighted individually, entirely new possibilities for dynamic processing and thus sound shaping of a group arise. To avoid distortion during compression, it is advisable toto smooth the gains determined from comparing the sum level with the corresponding threshold value. To avoid distortion, one embodiment provides that the signals representing the target amplifier values ​​determined in the channel gain calculation device (KVBE-m) are passed through a smoothing device before being used to drive the controllable output amplifiers. The smoothing can be implemented with a device similar to an envelope follower and can be defined by three parameters: As, Js, and Rs, which define an attack time As, a hold time H, s (English: hold) and a cooldown period A s(English: release) represent parameters that are individually adjustable, in digital processing as multiples of sampling cycles, and each defines a smoothing phase. Preferably, the same parameters are used for all audio control channels grouped together. Other smoothing configurations are also possible. In contrast to an envelope follower, the attack phase begins when the target gain is below the smoothed gain, whereas for an envelope follower, the attack phase begins when the instantaneous level is above the smoothed level. Particularly in a configuration where the audio signals are available in digital form as a sequence of values ​​sampled at equal time intervals, an electronic circuit in the form of an FPGA or a dedicated ASIC can be used. Applicant: STAGETEC GmbH 12.09.2025 Our reference: P18.The 918WO circuit allows processing to be performed at a significantly higher clock rate than the time-based sampling of the audio signal during digitization. This enables the implementation of circuits that, for example, execute specific calculation steps for an audio signal, which can then be used iteratively for the various audio control channels. Alternatively, the calculation can be performed simultaneously, i.e., in parallel, in all control channels. Processing in the compression control unit is preferably carried out in stages, since digital processing in processing sections or stages reduces the complexity of the calculation and facilitates the determination of values, for example, by allowing results or intermediate results to be used in one or more subsequent processing stages.One embodiment therefore provides that the determination of the quantities takes place in a multi-stage iterative or multi-stage parallel process, wherein in one iteration one or more of the quantities and / or one or more intermediate results are determined for one of the audio compression channels (AKK-m), wherein different quantities or different intermediate results can be determined simultaneously for different audio compression channels (AKK-M) in one iteration, and wherein the iterations of a stage are executed a number corresponding to the number of multiples M of the audio compression channels (AKK-m). In a particularly preferred embodiment, the multi-channel compressor is designed to also take panning of the audio signals output from the multi-channel compressor into account during compression. During panning, the audio signal of a channel is processed with different proportions (here referred to as panning factors) Pm. 1, Pm 2, …, PmN, mixed on N output channels or panning tracks. In the stereo case, where N equals two, the panning factors are also referred to as PL(θm) and PR(θm), where an angle θm is called the panoramic angle and the panning is set individually for each channel. This is achieved by splitting the optionally weighted and optionally filtered audio control channel signals into N-tuples according to the panning factors in the audio control channel. For these tuple values, i.e., separately for each panning track, an N-tuple panning track sum level and an N-tuple panning track level are determined. The sum level and the channel level are then determined as the maximum corresponding tuple value, i.e., the maximum panning track sum level and the maximum panning track level of the respective channel. The remaining signal processing remains unchanged.A simple design is achieved by arranging the individual audio compression channels side by side and indexing them. For each audio compression channel, it can be decided whether or not it is linked to the leftmost audio compression channel (without loss of generality). The leftmost channel cannot be linked in this way. Alternatively, it could be decided whether each audio compression channel is linked to the audio compression channel to its right. In this embodiment, the rightmost audio compression channel cannot be linked further. Linking to the leftmost audio compression channel is equivalent to linking to the audio compression channel with the next lower (or alternatively, the next higher) index for indexed audio compression channels.Especially with analog compressors, the linking device allows connections to an adjacent audio compression channel to be opened or closed. This groups the audio compression channels, which are linked and connected to each other via lines. An audio compression channel that is not linked to its left (or right) neighbor, i.e., not connected to another channel, starts a new group from left to right (or right to left). If it is not connected to the audio compression channel located to its right (or left), it forms a single-channel group. In this way, a varying number of group compressors (with automatic or fixed thresholds) or automixers can be configured. Each group can use a different compression mode.In one embodiment, the audio compression channels (AKK-m) are indexed, and the linking device allows each audio compression channel (AKK-m) to be linked or not linked to an adjacent audio compression channel (AKK-m-1) with respect to indexing. Applicant: STAGETEC GmbH 12.09.2025 Our reference: P18.918WO All audio compression channels (AKK-m) can be linked either to an audio compression channel (AKK-m) with a lower index or all only to an audio compression channel (AKK-m) with a higher index. The fact that an audio compression channel cannot be linked further indicates that an active, further linking is not possible. However, the audio compression channel can very well be linked to its neighbor or even further neighbors on the other side via an active linking of the other audio compression channel. For example, the first...The channel cannot be linked to a non-existent 0th channel on the left, but can be linked to the 2nd channel (and optionally further channels) via an (active) link between the 2nd channel and the 1st channel. To route audio signals through the multi-channel compressor without compression, the audio compression channels preferably have an activation device, for example in the form of a switching device such as a push button, toggle switch, or similar. If the audio compression channel is not active, the output gain is not reduced. In a particularly preferred embodiment, a group compression mode with automatic threshold setting can be selected in addition to or as an alternative to one of the two compression modes described above.Unlike the “normal” group compression mode, no constant threshold is used for determining the gain value, but rather the maximum channel level of the audio compression channels grouped together plus a positive offset ε is used as an adaptive threshold, but preferably limited to a range [T. u , T o], which is limited by a lower threshold Tu (lower limit threshold) and / or an upper threshold To (upper limit threshold). This compression mode is suitable, for example, for situations where audio signals from instruments, but also human voices, are grouped together, have been individually pre-compressed as needed, i.e., have already undergone signal processing within their channel strip, e.g., of a mixing console, before being fed into the multi-channel group compressor, so that the individual signals have controlled dynamics and do not require further compression. If the aim is now to level the group relative to itself due to fluctuating participation of active participants, the user's work can be made easier if the threshold is set automatically, i.e., adaptively.This is precisely what the group compression mode with automatic threshold achieves. Assuming that, depending on the passage of a piece of music, more or fewer audio signals are involved, i.e., they have a non-negligible signal value, especially a non-zero absolute value, then no compression occurs in passages where, for example, only one instrument or singer is active, since the threshold is always above its own level. This is ensured by the positive offset ε. When multiple voices or instruments enter, compression eventually kicks in, depending on the offset ε. From a musical perspective, orienting the compression towards the loudest audio signal seems plausible. Firstly, this signal often has the highest signal-to-noise ratio; secondly, it can temporarily dominate the overall mix and is therefore advantageously used as a level reference.Assuming that the individual audio signals are mixed together via individual faders before being fed to the multi-channel compressor, and that neutral weights are applied, the dominant signal, which is louder than the rest of the audio signals in the group, would eventually mask the other signals in the same group. At that point, the compression would be reduced because the threshold value relative to the overall level has automatically increased when the other audio signals no longer contribute significantly. However, if the group signals are all approximately the same volume, then the level of virtually every currently active signal, including the level of the loudest signal in the group, provides a good reference point for setting the compression gain.In such a case, the group compressor behaves approximately the same regardless of the absolute level of the individual signals, and its level reduction increases with the number of similarly loud signals involved. This allows the overall group to be balanced in a controlled manner, as mentioned. If there is also a positive weighting of a signal, for example, from a solo or lead singer, the reference point can be enforced. Vocal doubling then triggers the group compression depending on its level. The upper limit T. o This can optionally be used to prevent the overall volume of the group from becoming too high. The lower limit T uApplicant: STAGETEC GmbH 12.09.2025 Our reference: P18.918W Optionally ensures that no compression occurs in quiet passages with low overall group volume, so that the group is not masked compared to other signals in the mix. Level determination can be performed in different ways. A unit for determining a level is also called a detector, and the determination of the level is called detection. A fundamental distinction is made between level determination based on an amplitude value or the signal power. In the first case, the absolute values ​​of the audio signal, each plus a very small positive offset, are used for determination (peak value detection). In the second case, the squares of the absolute values ​​of the audio signals, optionally increased by another small offset (power data), are used. There are different variations for the different detection methods, as explained in more detail below.Preferably, the determined values ​​are logarithmized with respect to an arbitrary base. These logarithmic values ​​are then referred to as levels. The gain values ​​are then also calculated using these logarithmic values. Different methods can be used for this, which are referred to as the soft-knee and hard-knee variants. In the hard-knee variant, the compression, i.e., the reduction of the target gain, begins abruptly. The function curve exhibits a kink at this point. In logarithmic space, the two sections are linear. In the soft-knee variant, this kink region is replaced by a section of a polynomial, where the function values ​​of the polynomial function and the first derivatives of the polynomial at the boundaries of the replacement region are replaced by the function values ​​of the first derivative and the second derivative, respectively.The derivatives of the two linear sections of the gain function coincide, each being a function of the difference between the sum level and the corresponding threshold. In the hard-knee variant, the target gain, i.e., the logarithmic gain Gm(n), is equal to an optional negative impact factor I multiplied by the maximum of the difference between the sum signal level Sp(n) and the threshold Tm on the one hand, and zero on the other. This means that the logarithmic target gain Gm(n) is zero as long as the sum level Sp(n) is less than the threshold, and otherwise is the difference between the sum signal and the threshold, possibly multiplied by the optional negative impact factor. The impact factor I, also referred to as the effect factor or influence parameter, lies between 0 and 1. Applicant: STAGETEC GmbH 12.09.2025 Our reference: P18.918W When the logarithmic gain Gm(n) is zero, there is no attenuation of the output gain. The larger the absolute value of the logarithmic gain, which is always negative, the stronger the compression. The logarithmic gain is converted back into a linear target gain value by exponentiation before optional smoothing of the gain. The invention is explained in more detail below with reference to the drawings. Here, Fig. 1 shows a prior art signal processing system in which a bus signal formed from three signals is compressed and then reverb effects are added; Fig. 2 shows a prior art signal processing system in which reverb effects are added to a bus signal formed from three signals before compression; Fig. 3 shows a schematic representation of a multifunctional multi-channel compressor; Fig.4 Signal flow diagram of a detector for a single dynamic processing of a group of signals; Fig. 5 Schematic representation of an exemplary configuration of a multifunctional multi-channel compressor with two group mixes, an automix, and single compression; Fig. 6 Schematic representation of embedding the automixer / compressor in a flexible signal processing path; Fig. 7 A schematic representation of a sequential processing architecture and a timing scheme for processing M channels; Fig. 8 A schematic organization of a memory for signal processing, corresponding to the configuration according to Fig. 5; Applicant: STAGETEC GmbH 12.09.2025 Our reference: P18.918WO Fig. 9 A schematic representation of a control panel of a console system with group processing unit; Fig. 10 A schematic representation of an automixer or group compressor as a chain of a single basic element (U1, U2, …); Fig.Fig. 11 Schematic diagram for a basic element for dynamics processing according to variant 1; Fig. 12 Schematic diagram for a basic element for dynamics processing according to variant 2; Fig. 13 A schematic representation to illustrate its use of a multifunctional multi-channel compressor for stereo processing of mono and / or stereo channels distributed in the panorama. In Fig. 1, a signal processing device 1 according to the prior art is shown, in which signals from three channels 11, 12, 13 are combined in a summing amplifier 20 to form a bus signal 30. This bus signal 30 is fed to a group compressor or bus compressor 100, which generates a compressed bus signal 110. A so-called send channel 41, 42, 43 is branched off from each of the three channels 11, 12, 13. Each of the transmitting channels 41, 42, 43 includes a weighting device 51, 52, 53, which is designed, for example, as an adjustable amplifier.The signals from the three send channels 41, 42, and 43 are combined in a further summing mixer 60 to form a summed signal 70, which is fed to an effects unit 80. An effect signal 90 output from the effects unit 80, for example, a reverb unit, is combined with the compressed bus signal 110 in yet another summing mixer 120 to form the output signal 130. The three channels 11, 12, and 13 represent, for example, audio tracks of a miked drum kit, consisting of the kick, snare, and hi-hat tracks, which are pre-summed to form a group signal, also called a bus signal 30, i.e., a separate unit or "group." This typically means routing a specific selection of tracks to a mono, stereo, or surround bus that represents the group. Such a bus signal 30 can then be subjected as a whole to the following steps, as in the example shown, a joint compression of the one applicant: STAGETEC GmbH 12.09.2025 Our reference: P18.918WO The drum track representing bus signal 30 is used to limit the dynamics. This often results in a subjectively perceived compression or compression of the signal, which is a desired effect. However, this state-of-the-art processing presents a problem: applying desired effects to individual or all signals—for example, adding a separate reverb component to a single signal like the snare or adding a reverb component to all signals—leads to undesirable effects. If send channels 41, 42, 43 or send outputs are used partially for this purpose, the ratio between the dry and reverberated signal varies during signal processing according to Fig. 1, since the compressor modulates the volume of the dry signal, which is included in the sum. This varying wet / dry ratio is often undesirable. Fig. 2 shows another possible state-of-the-art signal processing method to circumvent this disadvantage.In contrast to the embodiment shown in Fig. 1, the effect signal 90 is combined with the bus signal 30 before group or bus compression in the bus compressor 100 and in the further summing amplifier 120, which is arranged before the bus compressor 100 and generates the output signal 130. Identical technical features are indicated by the same reference numerals in all figures. This variant has the disadvantage that the effect unit 80 could not be used for other sources outside the group and that the effect signal 90 itself is compressed, which is not necessarily desirable if a particularly authentic spatial impression is to be created. Both signal processing approaches are not entirely satisfactory depending on the requirements. Fig. 3 schematically illustrates a proposed multifunctional multi-channel compressor 200 and a signal processing method that can be implemented with it.The problems associated with the current state of the art can be circumvented by subjecting the individual tracks 211, 212, and 215 to a common dynamic processing, but without first combining them into a group signal. This allows the individual signals to remain available after dynamic processing, i.e., compression, and to be accessed via send channels. Applicant: STAGETEC GmbH, September 12, 2025. Our reference: P18.918WO. The multifunctional multi-channel compressor 200 comprises M audio compression channels: 210, 210-1, 210-2, ..., 210-M. Each of the audio compression channels 210, 210-1, 210-2, ..., 210-M includes an input 220, 220-1, 220-2, ..., 220-M. Up to M signals are fed into the multi-channel compressor 200, originating, for example, from M channel strips 180, 180-1, 180-2, 180-M of a mixing console. These channel strips 180 are indicated by the output faders 190, 190-1, 190-2, … 190-M.Each of the audio compression channels 210, 210-1, 210-2, …, 210-M further comprises an output 250, 250-1, 250-2, …, 250-M, as well as a controllable output amplifier 240, 240-1, 240-2, …, 240-M arranged between them. This is, for example, designed by means of a voltage-controlled amplifier (VCA). The controllable output amplifiers 240, 240-1, 240-2, …, 240-M are also referred to as compressors. The multi-channel compressor 200 comprises a compression control device 300, which includes an audio control channel 310, 310-1, 310-2, …, 310-M for each audio compression channel 210. An audio control channel 310 comprises an input 320, which is connected to the corresponding input 220 of the corresponding audio compression channel 210. Each audio control channel 310 further comprises, in the illustrated embodiment, a side channel filter 330, 330-1, 330-2, …, 330-M and a weighting device 340, 340-1, 340-2, …, 340-M.A channel gain calculation device 350 determines a target gain signal 390 as a control signal for each audio compression channel 210, 210-1, 210-2, …, 210-M or its controllable output amplifier 240, 210-1, 240-2, …, 240-M. The compression control device 300 summes all optionally filtered and optionally weighted audio control channel signals 360, 360-1, 360-2, …, 360-M of the audio control channels 310, 310-1, 310-2, …, 310-M in a summing amplifier 355 to form a summed control signal 370. The compression control device 300 further comprises at least one detector device 380, which, starting from the summed control signal 370 and the optionally filtered and optionally additionally or alternatively, weighted audio control channel signals 360, 360-1, 360-2, …, 360-M determine a preferably logarithmic sum level 375. Depending on the compression mode, the applicant: STAGETEC GmbH 12.09.2025 Our reference: P18.918WOD detection device 380 also for each audio control channel signal a preferably logarithmic channel level 365, 365-1, 365-2, …, 365-M. The sum level 375 and, if applicable, the channel levels are fed to the channel gain calculation unit 350, which generates a target gain signal 390 in a group compression mode. The value of this target gain signal indicates the gain for the corresponding audio compression channel 210, 210-1, 210-2, ..., 210-M. In principle, various compression modes are possible, so other embodiments may include a compression mode selection device (not shown). The individual target gain signals 390-M control the compression by means of the controllable output amplifiers 240 in the individual audio compression channels 210. The embodiment of the multi-channel compressor in Fig. 3 can be implemented using analog or digital technology.The diagram is to be understood only as a signal flow diagram and has no direct implications for the implementation. M input channels pass through their channel strips 180 (optional, not shown) and corresponding output faders 190 before entering the multi-channel compressor 200. In the summing amplifier 355 of the compression control unit 300, a weighted "virtual sum," the sum control signal 370, is formed from optionally pre-filtered channels or audio control channel signals 360, which is fed to the detection unit 380. The compression control unit 300 controls M parallel digital or analog output amplifiers 240 (voltage-controlled amplifiers = VCAs) via corresponding control signals, the target gain signals 390. As with all audio compressors, the dynamic range is limited by reducing the output gain as soon as a signal, in this case a group compression, exceeds the virtual sum, i.e.,The sum control signal 370 exceeds a threshold, possibly gradually. Since the individual signals, i.e., the output signals 260, 260-1, 260-1, …, 260-M of the audio compression channels 210, 210-1, 210-2, …, 210-M, remain as such, they can be used for send mixes 510, 510-1, 510-2, …, 510-M. In the example shown, individual channel signals, weighted via optional weighting devices 520, 520-1, 520-2, ..., 520-M, are sent to an effects unit (Applicant: STAGETEC GmbH, September 12, 2025, Our Ref: P18.918WO550), whose output signal 560 is summed together with a bus signal 270 in a master bus summer 290 to form a master bus signal 295. The bus signal 270 is previously summed from the output signals 260, 260-1, 260-1, ..., 260-M of the channels in a bus summer 280. The addition of weighting allows the influence of individual signals on the compression of the group to be increased or decreased.An example would be a relatively loud bass drum, whose influence is reduced by weighting it below 0 dB. Since each signal can be weighted individually, entirely new possibilities for dynamic processing and thus sound shaping of a group arise. Weighting represents an alternative or supplement to the use of the usual so-called sidechain filters. These are also included in the arrangement (optionally) as "SCF" blocks 320, which further expands the sound shaping possibilities. With regard to the bass drum example, its influence can also be achieved by attenuating the bass frequencies using a filter, leading to slightly different sonic results. In addition to the summed signal 370, the pre-weighted and filtered audio control channel signals 360 and 360-m, as shown, also go directly to the detection unit 380. This unit determines a channel level.They are required for an automix mode and a novel, self-inventing special compression mode with automatic six-point threshold, in which the threshold is determined automatically. This method is explained in more detail below. The terms "virtual bus sum" and "virtual bus compressor" refer to the fact that such a "bus sum" and such a "bus" exist only in the sidechain path and ensure that the dynamic processing behaves in the same way as with summation according to Fig. 1. The method is therefore not to be confused with virtual, software-modeled audio effects. Multi-channel bus compressor as a modification of an automixer. It can be seen that the signal flow shown in Fig. 3 essentially corresponds to that of an automixer, but performs compression as described. Applicant: STAGETEC GmbH 12.09.2025 Our reference: P18.918WO An automixer is a well-known device whose function, in short, is to distribute the gains of competing signals, such as speakers in a talk show, using normalization so that the overall gain of all signals, or the overall volume, remains constant, and the loudest channel receives the largest share of the gain. Passive microphone signals are typically suppressed during the presence of a relatively loud microphone signal from an active speaker to such an extent that the amount of ambient noise picked up by these open microphones decreases noticeably. Furthermore, the automixer increases the maximum gain of a connected loudspeaker system that is possible before feedback occurs. In addition to the weighted sum, the channel signals themselves are also required for the detector of the automixer. Suppression of bass or...The pre-filtering reduces intermodulation distortion in low mid-range frequencies, which arises from the control process with its finite response times. In contrast to the previous consideration regarding high-energy, loud signals (e.g., bass drum), the filters thus fulfill a function that weighting alone does not. If a digital signal processor is implemented using FPGA technology or even as a custom integrated circuit (ASIC) and includes an automixer, then chip area is typically allocated to this processing unit. The multifunctional multi-channel compressor has the advantage that, by modifying its functionality, it can implement both a multi-channel compressor and an automixer without significant additional effort. Therefore, the functionality of the automixer and the multi-channel compressor will be defined mathematically in the following section.Mathematical description of the method of a multi-channel virtual bus compressor, either on its own or as an automixer. The following definitions are made for discrete-time digital systems, which, due to technological advantages, also represent the most common application today. An implementation in analog technology will be presented later. When "time-invariant" coefficients or constants are mentioned below, this means that no time dependencies exist as long as these coefficients are not changed directly or indirectly by the user. Assume a discrete-time sampling system with a constant sampling rate fT = 1 / T, where T denotes the sampling period. For each sample value x(nT) of a signal in the system at time nT, an integer time index n suffices. Therefore, x(n) is simply written instead of x(nT).Equations are generally abbreviated as "Eq." and figures as "Fig." in the following. Equations are numbered consecutively using integers. For related equations, these are indicated by a common integer and a sub-numbering in the form of an incremental decimal place. Alternative equations that replace the corresponding originals are marked with a lowercase letter suffix, beginning with "b". In some cases, the alternatives receive their own numbers instead of a suffix, which is then explicitly indicated. Given M synchronous input signals, i.e., those subject to a sampling rate, aa. m(n), m=1…M of the arrangement to be defined in Fig. 3, where m denotes the channel index and n the time index. These input signals are usually post-fader output signals from channel strips on a mixing console. In the first version of the description, for the sake of simplicity, all M channels are assumed to be involved in the processing. As will be shown, unused channels can still be excluded from processing and pass through the arrangement without modification. The signal flow diagram above in Fig. 4 is a detailed view of a compression control unit 300 according to Fig. 3. Based on the following equations, the letters mark the signals of the same name after the individual process steps have been completed. Reference symbols are the same as those in Fig. 3. Preprocessing As shown, the input signals a m (n) initially pre-filtered optionally. For the output signal b m(n) of the m-th channel after passing through a linear, time-invariant, feedback filter of K-th order with the coefficients B k and A k As is known: Applicant: STAGETEC GmbH 12.09.2025 Our reference: P18.918WO The signals are then weighted by applying non-negative factors w m multiplied. ^^(^) = ^^(^) ∙ ^^ (2) A neutral weight corresponds to a weight of w=1. By setting its weight to zero, the influence of the m-th channel on the compression or automix can also be completely neutralized. Note that the filtering-weighting process order can also be reversed and lead to the same result. For both the automix mode and group compression mode, the sum of the preprocessed audio control channels 360 is required: ^(^) = ∑^^^^ ^^(^) (3) Detection The individual channels and the sum now each pass through a detector element 341, 341-1, 341-2, …, 341-M, 341-S with the aim of estimating either the peak value or the average power. In the latter case, the RMS value is determined later. The response times of the sum detector and channel detectors, resulting from coefficients, are chosen to be the same or similar, with values ​​suitable for practical applications. In the following, two heuristic variants (6.1) and (6.2) for peak value estimation and two further variants, (7.1) and (7.2), for average power are introduced. Each of these detectors is, in principle, an operator with memory in the form of a number of state variables, which provides an output value y(n) for a current input value x(n). First, define Applicant: STAGETEC GmbH 12.09.2025 Our Reference: P18.918WO^^(^) = |^(^)| + ^^ (4) xa(n) here corresponds to the absolute value of x(n) plus an optional, very small offset o1, which is introduced for numerical reasons. xp(n) is the power square, where a small, optional offset o2 can also be used. The first peak-value detector type determines the maximum of the last L rectified samples. ^^(^) = max {^^(^), ^^(^ − 1), … , ^^(^ − ^ + 1)} (6.1) The second peak-value detector type estimates the peak value using an envelope follower with a response time, hold time, and decay time, represented in that order by three coefficients Ae, He, Re, where He is an integer number of samples. The state of the detector consists of the previous output value y(n-1) and a counter value z(n-1). The current initial value and the new counter reading are determined as described below. Three cases must be distinguished. (6.2) ^(^) ≔ ^(^ − 1) = 0 for ^^(^) ≤ ^(^ − 1) and ^(^ − 1) = 0. During the attack phase, when the input value exceeds the detector state, the counter is set to the hold time. As soon as the input signal falls below the current estimate of the peak value after the attack phase, the hold phase begins, and only after the counter has completely counted down to zero does the release phase begin, unless the attack phase is triggered again. The attack time and hold time can also be set to zero. Note that y(n) cannot assume negative values, as the input current is always positive. The first type of power detector consists of the mean of the squares of the last L samples: The second type of power detector uses a feedback first-order filter with coefficient c0, where |c0|<1:^(^) = ^^ ∙ ^^^(^) − ^(^ − 1)^ + ^(^ − 1) (7.2) For the application, choose one of these four detectors and apply this type to the sum and the individual channels, where the coefficients used can vary slightly between the sum detector and the channel detectors. Let Dch{x(n)) denote the operator for the individual channels c m (n), which results from equations (4) and (5) and the detector selected from the palette (6.1), (6.2), (7.1), (7.2). Similarly, let Ds{x(n)} denote the operator according to the selected detector for the sum s(n). These memory-based operators operate not only on the current sample but also on past samples, i.e., the entire sequence. Then, for the individual signals in Eq. (2), a total of M detector signals d are obtained. m(n) and for the sum in Eq. (3) a summing detector signal se(n):^^(^) = ^^^{^^(^)} (8)^^(^) = ^^{^(^)} (9) Applicant: STAGETEC GmbH 12.09.2025 Our reference: P18.918WO When using the automix mode combined with one of the presented power detectors, the summing detector signal should be limited from below so that the sum does not become smaller than individual signals in the case of out-of-phase, correlated recordings. The limit is defined as the sum of the power contributions of the individual channels, which is formed at a summing amplifier 342: ^^(^) = ∑ ^ ^ ^^ ^ ^(^) (10) This limit corresponds approximately to the sum of the power of M uncorrelated signals. Then the sum detector signal is to be bounded from below, and instead of Eq. (9), the following shall apply: ^^(^) = max {^^{^(^)}, ^^(^)} (11) max{a,b} denotes here and in further equations the maximum value of a and b. In general, both in group compression mode and in automix mode, the task of the entire detector circuit 380 together with the channel gain calculation device 350 is to calculate a continuous, linear, “raw” target amplitude gain H for each channel m. m (n) to calculate, which is then smoothed. Without considering further factors that will be introduced, the following would have to apply when using peak value detectors, specifically for the automix mode: However, when using power detectors, the following would have to apply: As can be seen, the gain in automix mode is limited to a value of 1 (corresponding to 0 dB). In addition, there is the compression mode, which requires a different rule. Applicant: STAGETEC GmbH 12.09.2025 Our reference: P18.918WO The equations (12) and (13) above merely indicate how an automixer should behave, but are not used in this form in connection with the procedure. The calculation of the target gain H can instead be simplified and performed in the logarithmic domain. Any base β is suitable for this, and without loss of generality, the base β=2 is chosen here. A different base only changes the scaling. The detection device here comprises logarithmic elements 343-1, 343-2, …, 343-M, 343-S. The detector signals for channels and sum are given by the following expressions.When using peak values, which can also be called peak mode, it is possible to determine the logarithmic detector signals D independently of the mode (automix or compression). m (n) and S(n) write the following: ^^(^) = ^^^^(^^(^)) (14.1)^(^) = ^^^^(^^(^)) (14.2) When using power detectors, which can also be called RMS mode, effective values ​​are obtained by halving the logarithmic detector values, so that both detector types can subsequently be treated in the same way. Thus, the detectors operate as RMS detectors (RMS = root mean square). ^^(^) = ^^^^(^^(^)) / 2 (15.1)^(^) = ^^^^(^^(^)) / 2 (15.2) Note: D m(n) in Eq. (14.1) or (15.1) shall be called "channel level". S(n) in Eq. (14.2) or (15.2) shall be called "sum level". Applicant: STAGETEC GmbH 12.09.2025 Our reference: P18.918WO Target Gain Since the levels of the audio control channel signals and the sum of the audio control channel signals are in the logarithmic range, a logarithmic target gain G can be determined in the channel gain calculation device 350 with determining elements 351-m, which is transformed back into the linear range into the amplitude gain H by inverse transformation in power converters 352-m before smoothing is performed in smoothing elements 354-m. At this point in the determination of the logarithmic target gain G, the procedure now distinguishes between two different target curves, depending on whether the target gain as a function includes a "hard" or "soft" transition.Furthermore, the three operating modes are distinguished here: automix, group compression, and group compression with automatic (adaptive) threshold. "Hard-Knee" Variant: For the so-called "hard-knee" case, a target gain G is defined in the logarithmic range relative to the selected base β for each channel. m (n) (“gain”), a generalized threshold T m (n) (“threshold”) and an influence I m(“impact”) with the following relationship: ^^(^) = −^^ ∙ max{^(^) − ^^(^), 0} = −^^ ∙ F(^(^) − ^^(^)) (16) with^(^) = max {0, ^} (17) Due to the limitation in (16) using the “kink function” F(x), the target gain G can only take on negative values ​​or the value 0. As long as the sum S(n) remains below the threshold value, no gain reduction takes place (i.e., G=0). For the influence parameter I, 0 ≤ I ≤ 1 always applies. “Soft-Knee” variant Applicant: STAGETEC GmbH 12.09.2025 Our reference: P18.918WO Instead of approach (16), softer, more transparent curve transitions of the target gain can be achieved by using a polynomial P s(x) is used, which replaces the kink function F(x) in the right side of (16) within a chosen logarithmic transition region W = [-δ / 2, δ / 2] and operates on the difference x=ST. In this variant, the logarithmic gain is defined as ^^(^) ≔ −^^ ∙ P ^^(^(^) − ^ (^)) for −^ / 2 ≤ ^(^) − ^^(^) ≤ ^ / 2 (18.1)^^(^) ≔ −^^ ∙ F(^(^) − ^^(^)) for ^(^) − ^^(^) > ^ / 2 (18.2) or for ^(^) − ^^(^) < −^ / 2. Only within the range W, where (18.1) holds, does the curve in (18) deviate from Eq. (16). The polynomial Ps(x) used has at least second order and satisfies the property that the function value Ps(x) and the first derivative Ps'(x) at the transitions x=-δ / 2 and x=+δ / 2 coincide with the kink function F(x) or its derivative: ^^(−^ / 2) = 0 (19.1)^^(+^ / 2) = ^ / 2 (19.2) Operating Modes Now we can turn to the different operating modes. Another quantity is introduced: the maximum level within the channels at sample time n. The maximization is performed via the channel index m on a maximum finder 345. ^^^^(^) = ^^^^{^^(^)} (20) Another quantity is the following adaptive threshold, which results from the maximum level plus a non-negative offset ε and, optionally, an upper bound T. o and / or optionally lower barrier T u shall be subject to:^^(^) = ^^^{^^^{^^^^(^) + ^, ^^}, ^^} (21)Applicant: STAGETEC GmbH 12.09.2025Our reference: P18.918WOJe depending on the operating mode, the channel-related threshold T mIn Eq. (16) and Eq. (18), different quantities are assigned according to the following table: Table 1: Threshold for channel m depending on the operating mode Operating mode Automix Group compression Group compression with automatic threshold Threshold Channel level Threshold constant Adaptive threshold T m D m C Ta 1 . AutomixIn "Automix" mode, according to Table 1: ^^(^) ≡ ^^(^) (22) According to Eq. (16) or (18) and (22), the sum level is compared against the individual audio control channel level for each audio control channel. For the hard-knee variant (16), this corresponds in the linear range to the quotient in Eq. (12) for peak detection or Eq. (13) for RMS detection. However, the method is extended such that the logarithmic gain G can be adjusted, i.e., attenuated, using the impact parameter I, and that the soft-knee approach in Eq. (18) is available as an alternative. From the user's perspective, the louder the individual channel level is relative to the sum level, the less level reduction occurs. With only one signal in the sum, apparently no level reduction takes place.With a dominant signal in the form of an active speaker, weak signals are strongly suppressed, while the dominant signal itself is hardly affected, depending on the noise component. Pathological cases, where strongly correlated but antiphase signals are present in different channels and cause the sum level S(n) to be potentially lower than one of the channel levels involved, are countered by limiting the gain in Eq. (17). 2. Group Compression Applicant: STAGETEC GmbH 12.09.2025 Our Ref: P18.918WO In "Group Compression" mode, the varying channel level is replaced as the threshold by a threshold constant C. The following applies: ^^(^) ≡ ^ (22b) As can be shown by calculations, this results in dynamic processing that corresponds to that of a forward-facing compressor that uses the selected detector for the weighted channel sum, i.e., operates in peak or RMS mode.With respect to the usual compressor ratio R, the following simple dual relationship with the impact factor I can be derived for the hard-knee case: R = 1 / (1 − R) (23.1) R = 1 − 1 / R (23.2) As the impact factor approaches the value 1, the ratio R approaches infinity, resulting in a limiter. Generally, one has the choice between a global impact parameter I, which is the same for all channels, and individual impact parameters Im, an impact parameter I. mper channel. The latter case allows the influence of group-related compression to be set individually for each channel, giving each channel an individual ratio. This can be used, for example, to ensure that a lead vocal in an overall mix is ​​barely reduced in level under strong group compression, even though its contribution to the overall level S(n) may be significant. Note that S(n) for automixers and multi-channel compressors is a mono sum. A stereo version for the multi-channel virtual bus compressor is described below. The mono sum determines the compression behavior, even if the channels are subsequently panned and summed. Applicant: STAGETEC GmbH 12.09.2025 Our reference: P18.918WO3.Group compression with automatic threshold (adaptive threshold) In this case: ^^(^) ≡ ^^(^) (22c) This new and self-inventive mode differs from simple group compression in that the adaptive threshold T. a instead of the constant C. According to Eq. (21), T aFor each time point n, the current maximum channel level plus an offset ε is calculated, preferably limited to the range [Tu,To]. The relationship between impact factor and ratio is the same as for simple group compression, i.e., Eq. (23) applies. The basic idea of ​​this operating mode will be briefly explained. It is assumed that instruments or human voices to be grouped together have already been individually pre-compressed as needed, i.e., have undergone signal processing within their channel strip, so that these signals, considered individually, have controlled dynamics and do not require further compression. If the aim is to level out a group relative to itself due to fluctuating participation of active members, the user's work can be made easier if the threshold is set automatically, i.e., adaptively.Assume that, depending on the passage of a piece of music, more or fewer signals are involved. With a single signal, compression apparently does not occur, as its threshold is always above its own level. When additional voices are added, compression eventually sets in, depending on the offset ε. Orienting oneself towards the loudest signal according to (21) seems musically plausible. Firstly, this signal often has the highest signal-to-noise ratio; secondly, it can temporarily dominate the overall mix and is therefore advantageously used as a level reference. For a moment, assume that the tracks are indeed mixed together according to the fader positions and that neutral weights are present. If the dominant signal is significantly louder than the rest, other signals in the same group would eventually be masked.Then, however, the compression is reduced because the threshold value has automatically increased relative to the overall level if the other signals don't contribute significantly. If, on the other hand, the group signals are all roughly the same volume, then the channel level of virtually every temporarily active signal, including the channel level of the currently loudest signal in the group, provides a good reference point. In this case, the compressor behaves roughly the same regardless of the absolute channel level of the individual signals, and its level reduction becomes more pronounced the more (similarly loud) signals are involved. This allows the overall group to be balanced in a controlled manner, as mentioned. By positively weighting one signal, for example, a lead singer, the reference to this signal can be enforced. Vocal doubling then triggers group compression depending on its level.The upper limit To can optionally be used to prevent the overall group volume from becoming too high. The lower limit Tu, in turn, optionally ensures that no compression occurs in quiet passages with low overall group volume, so that the group is not masked by other signals in the mix. Potentiation and smoothing of the... The resulting logarithmic “crude” target gain G m (n) for the m-th channel is transformed by exponentiation to the chosen basis β=2 to obtain the linear gain H m to obtain (n). As mentioned before, choosing other bases yields equivalent results. ^^(^) = 2^ ^(^)(24) As with any conventional compressor, smoothing the target gain with asymmetric filtering is an essential part of the functionality to avoid significant distortion. The smoothing works similarly to the threshold detector in (6.2), i.e., with an attack time, hold time, and release time, in that order. Applicant: STAGETEC GmbH 12.09.2025 Our reference: P18.918WO represented by three coefficients As, Hs, Rs, where Hs is an integer number of samples. However, unlike the detector, the attack phase here begins when the linear target gain H is less than the smoothed gain J. Each smoothing element, in turn, has two states: the previous output value J and the m (n- 1) and the counter reading Z m(n-1). The cooldown is usually chosen to be significantly longer than the attack time. Attack: (25)^^(^) ≔ ^ ^ ^^ ∙ ^^ (^) − ^ (^ − 1)^ + ^^(^ − 1)^^(^) ≔ ^ ^^ for ^ (^) < ^^(^ − 1) Hold:^^(^) ≔ ^^(^ − 1) ^^(^) ≔ ^^(^ − 1) = 0 for ^^(^) ≥ ^^(^ − 1) and ^^(^ − 1) = 0. The described variant operates in the order of exponentiation-smoothing. For compressor mode, a reverse order is also possible, i.e., first smoothing in the logarithmic range and then exponentiation. For automix mode, set As = Rs and Hs = 0. This results in a first-order, feedback-controlled low-pass filter, which ensures that the smoothed gain does not exhibit abrupt jumps. Output Signals: The output signal ym(n) for the m-th channel, i.e., the audio compression channel, is the product of the input signal, the smoothed gain, and an optional catch-up gain V ≥ 1 for compressor operation. In this way, all M output signals are obtained. Applicant: STAGETEC GmbH 12.09.2025 Our reference: P18.918WO^^(^) = ^^(^) ∙ ^^(^) ∙ ^ (26)The input signal a m could also be in opposition to the reinforcement J mhave a time lead or lead in integer multiples of samples. This would not substantially change the method. In Eq. (26), time differences are not taken into account. Organization of single and multiple automixes and group compressions. In the method presented so far, all M channels are involved to obtain a single automix or a multi-channel, i.e., group, compression. The term "group processing" can be used here as a general term independent of the operating mode. A channel can be removed from the group processing to which it is assigned by setting its weight and its impact factor to zero. Several group processing operations can exist concurrently. In this case, the signal flow in Fig. 4 can simply be imagined as vertically multiplied, so that further inputs and outputs are added.This is legitimate insofar as an input channel is assigned to at most one group processing operation. Each duplication has its own channel sum. The numbering of the input channels is variable. The channels assigned to a group processing operation do not necessarily have to be located next to each other on the mixing console. Flexible, sequential implementation of the method: There are many possible implementations. The signal flow diagrams in Fig. 3 and Fig. 4 are to be understood as exemplary embodiments and define only one exemplary implementation. Applicant: STAGETEC GmbH 12.09.2025 Our reference: P18.918WO A slightly modified variant, which can be implemented particularly elegantly, is described below. As before, let M channels be given as inputs. This time, however, it should be possible to process channel bundles separately. An example configuration for this approach is shown in Fig. 5 for M = 12 channels.Audio compression channels 210-1 to 210-3 are subjected to a group compression 710-1, and audio compression channels 210-4 to 210-7 are subjected to a different group compression 710-2. Audio compression channel 210-8 is not processed at all, while audio compression channels 210-9 to 210-11 are components of an automixer 720. Audio compression channel 210-12 undergoes a standard (group) compression 710-3, with the group comprising only this single audio compression channel 210-12. Each group processing and individual processing operation should have its own set of parameters. They are therefore distinct from one another.Although they have separate signal processing capabilities, they can also include the same parameters in their own parameter sets. The limitation of this approach, to enable ease of use and / or simplify the calculation of individual signals, is that the channels belonging to a processing operation must be arranged adjacently on a user interface or be addressable with adjacent indices. An input routing matrix can be provided in the system, which, by reordering, ensures that physical inputs are fed to the desired adjacent channel strips. Such a system is indicated by way of example in Fig. 6, where the optional channel strips in the middle are not specified in detail.Incoming signals 810-1,… 810-M are rearranged in a router 820 so that the signals to be compressed together are fed onto adjacent channel strips, for example, of a mixing console 830, and then fed to a multi-channel compressor 200. This implementation uses a sequential processor which generates a vector of M output channel samples within one sampling period from a vector of M input channel samples with time index n. This processing is shown by way of example in Fig. 7. The delivery of the channels can, for example, be carried out using the time-slot method, where, as shown, the sample data arrives sequentially on a data bus at short intervals, is processed, and is output sequentially in the bundle. Applicant: STAGETEC GmbH 12.09.2025 Our reference: P18.918W. While the input data 910 is arriving, output data 920 can be output.During time slot 900(n), a bundle of sample data is processed 950. This principle is efficient, but not mandatory. A processor 930 has an internal or external memory 940 of any configuration for storing intermediate results, states, and the like. To implement flexible channel partitioning for different group processing, partial sums must be calculated instead of a global sum s(n) according to Eq. (3). The same applies to the sum of the power contributions su(n) according to Eq. (10) if an RMS detector is used in automix mode. Likewise, determining a maximum according to Eq. (20) only extends to those audio compression channels that are included in the maximum calculation, i.e., to the audio compression channels of a group that requires this maximum, calculated from the set of its channels, as its size.Simple two-step procedure for determining partial sums and partial maxima. Partial sums are defined in this context as sums of channels of a group. The following approach can be chosen to determine such partial sums: A binary operation vector v. m is defined such that v m = 1 should hold if and only if channel m is "coupled" to the previous channel m-1, i.e., this and the previous channel both belong to the same group processing, and v m= 0 should then apply if these channels belong to different processing operations, i.e., groups. With this definition, v1 for the first channel m=1 is always equal to 0, since it has no previous neighbor. The following pseudo-code can be executed on a system to calculate arbitrary non-overlapping partial sums, so that ultimately an example configuration as in Fig. 5 can be implemented. Pseudo-code 1: Summation and distribution of the partial sums Iteration 1: Applicant: STAGETEC GmbH 12.09.2025 Our reference: P18.918WO For all indices m from 1 to M, execute (loop) if v m = 0 (new sum, since there is no predecessor) set a := x m (first addend) set i := m (remember base index i) set Ki := a (store intermediate result in cell i) else a := a + x m(accumulate a with the next component) set Ki := a (store intermediate result in cell i) End of case distinction End of loop Iteration 2: For all indices m from 1 to M execute (loop) if v m = 0 (new sum is due, since there is no predecessor) set a := K m (fetch stored complete sum) End of condition set ym := a (store sum in output vector at index m) End of loop In the code above, x m An input vector, which in turn can be a memory area. Then there are the temporary variables or registers a (data value or accumulator), m (running index), and i (base index). K mis a vector with M elements of some memory area, whose elements are addressed using index m. Step 1 is used to determine the (partial) sum of each group, while step 2 serves to distribute the sums to all relevant elements of the associated groups. After both steps have been completed, all partial sums have been determined. Applicant: STAGETEC GmbH 12.09.2025 Our reference: P18.918WO Note: In iteration 1, data is temporarily stored after each step (cell i). This is not strictly necessary, as only the last summation of a partial sum needs to be stored. Partial maxima of an input vector can also be determined in a similar manner: Pseudo-code 2: Determining maxima of individual groups Iteration 1: For all indices m from 1 to M, execute (loop) if v m = 0 (no predecessor → new maximum determination) set a := x m(first value) set i := m (remember base index) set Ki := a (store intermediate result in cell i) else if a < x m (new, temporary maximum found) a := x m (Record new maximum) set Ki := a (store this intermediate result in cell i) End of case distinction End of case distinction End of loop Iteration 2: For all indices m from 1 to M execute (loop) if v m = 0 (new group is coming up) set a := K m(retrieve stored maximum) End of condition set ym := a (store current maximum in element m) End of loop Applicant: STAGETEC GmbH 12.09.2025 Our reference: P18.918WO Step 1 is used to determine the (partial) maximum of each group, while step 2 serves to distribute each maximum to all relevant elements of the corresponding group. Inclusion of partial sums and maxima in the procedure The following steps are now carried out to implement a partitioning of the channels for different group processing. First, the intermediate results c m (n) are determined in equation (2) and are stored as data in memory. As a reminder, this data represents the current signal values ​​after filtering and weighting. To extend the originally outlined procedure, the sum s(n), defined in equation (3), is now replaced by a vector of M sum values ​​s. m(n) replaced. The component s m (n) here represents a sum applied exclusively to channel m. To determine the partial sums, the two-step procedure in Pseudo Code 1 is used above, where cm(n) takes the role of the input vector with elements x. m slips, i.e. x m (n) := c m (n), and the output vector y m the sums s m (n) results in, therefore s m (n) := y m (n). The retention of the time index indicates that this two-step process is repeated with each sample n. It is noted that only a single group processing is performed for all audio compression channels, all s m (n) are equal and s(n) correspond as before. In the case of multiple group operations, the sums of the elements of a common group are identical. The lower bound s proceeds in the same way. u(n) from Eq. (10), which is only needed for the automix mode in combination with RMS detectors, as a total sum into partial sums s u m (n) over. Finally, the maximum of the channel levels is also determined group-wise according to Eq. (20) using Pseudo Code 2 and then yields D max m (n), is therefore an individual value for each applicant: STAGETEC GmbH 12.09.2025 Our reference: P18.918WO Audio compression channel, where in turn all these maxima are the same within a common group. The introduction of the sum vector s mThis implies that all sum components must be detected individually and then transformed into the logarithmic domain, which increases the computational effort accordingly. With the described adjustments regarding channel-specific sums and maxima, and applying these to all corresponding equations, Eq. (16) for the hard-knee case becomes: − ∙ max{ − ∙ F( − ∙ ... for ^^(^) − ^^(^) > ^ / 2 (18.2b) or for ^^(^) − ^^(^) < −^ / 2 The automatic threshold in Eq. (21) is expressed by ^ ^^ (^) = ^^^{^^^{^ ^^^^ (^) + ^, ^^}, ^^} (21b) As already indicated, this results in a significant increase in effort, but it offers the possibility of running many group processing operations in parallel through partitioning. With M unlinked channels, up to M parallel mono compressors can also be configured using manual thresholds. Figure 8 illustrates the described data organization using the example in Figure 5. For the linking vector vm = (0,1,1,0,1,1,1,0,0,1,1,0), the virtual channel sum s m (sum), the lower limit su m (“sum bound”), the sum level S m (sum level) and the maximum channel level Dmax m(“max channel level”) listed. s1, s2,…, u1, u2,…, S1, S2,…, M1, M2,… are any numerical values. The crucial factor is which channels are assigned the same values. Operating and Display Concept One possible, though by no means the only, way to operate this multi-channel compressor will be presented below. It is assumed that the multifunctional multi-channel compressor is part of a mixing console system. Fig. 9 shows an operating section of a console system 1000 with a group processing unit in the form of a multifunctional multi-channel compressor. It is assumed that only adjacent channel strips or adjacent channels can be linked to groups for group processing.There is either a physical control unit in the form of parallel channel strips 180-1, … 180-M with their controls in the form of faders 190-1, …- 190-M, or a virtual mixing console view on a screen. Each channel strip 180-1, … 180-M comprises two control knobs 205-1, …, 205-M; 206-1, …, 206-M: One control knob 205-1, …, 205-M, referred to here as the activation button, activates or deactivates the corresponding audio compression channel in the multi-channel compressor for group processing. When deactivated, its influence I and weight W should be set to zero. The corresponding audio signal passes through the audio compression channel without compression. The other button, 206-1, ..., 206-M, the link button, links each corresponding audio compression channel to its left neighbor (Variant 1) or, alternatively, to its right neighbor (Variant 2). The choice of variant is irrelevant here.A set of link buttons essentially corresponds to the physical representation of a link vector in Table 2. As soon as a link exists between two adjacent audio compression channels, group processing is already in place. Further audio compression channels can be added by chaining them using their link buttons. This can be indicated in a virtual view, for example, by framing the linked channels or changing the background area or its color. A separate view in the form of an embedded window or another page in the screen display must exist to define the operating mode and all relevant parameters. This can be done either by selecting the aforementioned frame or area, or by selecting a position of a selection button that forms a compression mode selection device. In Fig.Channel 9 shows three middle channel strips, 190-3 to 190-5, grouped together for processing by two left-facing Link buttons, 206-4 to 206-5. Signal processing is activated for all three channel strips, 190-3 to 190-5, in the audio compression channels 210-3 to 210-5. The activation buttons, 205-3 to 205-5, are active (black button position). The channel strips are shown schematically only with their output faders 190, 190-1, ... 190-M, and not with their full range of functions. This operation allows adjacent channels to be spontaneously grouped and subjected to common dynamic processing in one of the operating modes. The user should be able to quickly grasp the configuration if the channels are labeled accordingly. The arrangement of the control buttons 205 and 206 can be arbitrary.Alternative Implementation Based on Parallel Identical Processing Units in Analog or Digital Technology. In the last section, a simple two-step method for determining partial sums was introduced. This section presents another implementation variant that can also be realized in analog technology. The underlying principle is to build the multifunctional multi-channel compressor 200 as a chain of parallel processing units as basic elements 1100 (Fig. 10) to form one or more automixers and / or group compressors. This is described below starting from an analog signal path. The functionality of the chained basic element 1100 corresponds mathematically exactly to the method described in the last section. Arrows pointing outwards at the connections indicate the presence of an active driver stage.Each basic element 1100-m comprises an input 220-m and an output 250-m. Furthermore, there is a logic input 221-m for a binary logic signal, which indicates whether the corresponding basic element is linked to an adjacent basic element or not. The incoming logic information 1103-m is also communicated to the left-adjacent basic element 1100-m-1, as indicated by the dashed arrows. In the representation in Fig. 10, it should be noted that basic element 1100-1 has no left-adjacent element. A basic element 1100-M (not shown) accordingly has no right-adjacent basic element. Each basic element 1100-m outputs a forward sum 1140-m via a forward sum output 1142-m to a forward sum input 1141-m+1 of the next linked right-adjacent basic element 1100-m+1.Likewise, it receives a reverse sum 1150-m+1 from the adjacent basic element 1100-m+1 at a reverse sum input 1151-m, which the latter outputs at a reverse sum output 1152-m+1. Furthermore, all basic elements 1100 are linked to a common potential rail 1200, which is raised to a maximum potential by the basic element 1100 detecting the highest level. This maximum potential corresponds to the highest level detected at all basic elements 1100 in the group. Each basic element 1100-m thus has, in addition to the input 220-m and the output 250-m, a potential rail input 1201-m and a potential rail output 1202-m, a forward summation input 1141-m and forward summation output 1142-m, as well as a reverse summation input 1151-m and a reverse summation output 1152-m. Various configurations are possible, two of which are explained in more detail here. Variant 1 Applicant: STAGETEC GmbH 12.09.2025 Our reference: P18.918WO A first variant is shown in Fig. 11. A basic circuit element 1100 with index m is depicted. This element has a link information 1103-m indicating whether this element and the following one are linked to their left neighbor or not. If so, the corresponding link switches 1145-m are closed. The forward sum m-11140-m-1 includes all filtered and weighted, i.e., pre-processed, signal components of the left side. If the current element m 1100-m is linked to the left (link bit m is logic "1"), its optionally weighted and / or optionally filtered channel signal m 1110-m is added to the received forward sum 1140-m-1. This takes place in the summing circuit 1130-m and results in the forward sum 1140-m that is passed on. A filter and a weighting device are indicated in block 1120. This results in the forward sum m 1140-m, which is then passed on to the right neighbor 1100-m+1.If the current element m 1100-m is not linked, forward sum m 1140-mund and the current, pre-processed channel signal m 1110-m are identical, i.e., the forward sum m-11140-m-1 is not added. This is controlled by a switch 1145-m, which is switched depending on the link information 1103-m. If the link information 1103-m is zero / false, there is no left-hand link and the switch 1145-m is open. If another basic element m+11100-m+1 is connected and linked (link bit m+1 is logic "1"), a reverse sum m 1150-m corresponds to the reverse sum m+1 1150-m+1, since a corresponding further switch 1155-m selects this sum (link bit m+1 is logic "1"), which is switched depending on the link signal forwarded by the right basic element 1100-m+1 or the link information 1103-m+1.The further switch 1155-m is in position B if the linking information of the right-adjacent basic element 1100-m+1 is true (linked to the basic element m), otherwise in position A (basic element 1100-m is the last basic element 1100 on the right of the grouping). The backward sum m+11150-m+1 includes not only the sum of the right-hand side, but of all linked audio compression channels, i.e., the left, right, and current audio compression channels, because the last chained basic element 1100 on the right-hand side has the entire sum, and this sum is passed back recursively as the backward sum 1150. Applicant: STAGETEC GmbH 12.09.2025 Our reference: P18.918WO. The backward sum m+11150-m+1 is therefore equal to the previously introduced virtual total sum s. m1350-m and is therefore routed to a detector 1160-m, which calculates a logarithmic level 1170-m. If no further circuit element m+11100-m+1 is connected, the end is reached, and the sum of the forward sum m-11140-m-1, if present and linked, and the current pre-processed channel signal 1110-m is equal to the backward sum m 1150-m and, as already mentioned, also simultaneously the total sum s mThe summing amplifiers 1130-m shown can be implemented with known operational amplifier circuits. Furthermore, there is a potential rail 1200 for all linked basic elements 1100, which the audio compression channel 210 with the highest level can pull to its maximum value. Known driver circuits with diodes can be used for this purpose; in the illustrated embodiment, this is part of a compression channel detector 1195-m, which determines the channel level 1210-m. The adaptive threshold 1220 is derived from this maximum of all linked channel levels 1210. There is also a further switch 1205-m, which depends on the linking information 1103-m of the basic element 1100-m. Ensures the correct linking of the basic elements 1100 in the grouping with the potential rail 1200. Channel gain calculation device 1250-m calculates the target gain for the audio compression channel 210.The calculation incorporates the logarithmic level 1170-m of the summed signal and, depending on the compression mode, a threshold value 1260-m, as well as an optionally selectable / adjustable influence factor 1270-m. A selection circuit 1240-m chooses the appropriate value for the threshold value 1260-m depending on the operating mode. In "normal" group compression, this is a preselected / adjustable constant threshold value 1280-m. In an automix mode with an adaptive threshold, it is the maximum channel level 1218 added with an offset 1290-m. In automix mode, it is the detected channel level 1210-m. The output amplifier 240-m is controlled by the target gain signal 1258-m. The lower limit sum 1340-m for the automixer in RMS mode is only indicated in the circuit diagram for space reasons and can be summed according to the same principle. Applicant: STAGETEC GmbH 12.09.2025 Our reference: P18.918WO as well as the virtual sum sm 1150-m.The measured power is summed and serves as a power sum limiter. The summation can be performed analogously to the forward and reverse summation, so that only linked channels are summed. Variant 2: As described, Variant 1 in Fig. 11 requires the linking information 1103-m+1 of the subsequent element m+11100-m+1. Variant 2 in Fig. 12 can dispense with this information and the corresponding bit line, but requires three summing elements 1130-m, 1180-m, 1190-m instead of one. In this variant, the reverse summation m+1 1150-m+1 corresponds only to the sum of the channels on the right-hand side and not, as before, to the overall sum. If no further element 1100 is connected, this reverse summation 1150-m+1 is equal to 0 (no signal), for example, by pulling the potential to signal ground via an electrical resistor.The reverse sum m 1150-m is obtained, as shown, simply by summing the current pre-processed channel signal m 1110-m and the reverse sum m+11150-m+1 at the summing unit 1180-m, if a left link exists (link bit m, i.e., the link information 1103-m is logically "1"). The forward sum m 1140-m is obtained in the same way as in variant 1, while the final sum s. mThe summation is formed by summing the forward sum m 1140-m and the backward sum m+11150-m+1 at the summing mixer 1190-m, since this forward sum m 1140-m includes all relevant left audio compression channels 210-i (i≤m) including the current audio compression channel m, and the backward sum m+11150-m includes all relevant right audio compression channels 210-k (k>m), starting with m+1. In this variant 2, the connections via the switches 1145-m, 1155-m, 1205-m with the left basic element m-1 depend on the connection information 1103-m. In both variants, the illustrated gain block includes the calculation of the target gain and the smoothed gain, which is finally applied to the output amplifier 240-m, designed as a VCA. This part applicant: STAGETEC GmbH 12.09.2025Our reference: P18.918WO has already been fully described from a mathematical perspective in the previous sections.Stereo variant of the multi-channel compression method with envelope detector (peak detector). So far, only mono sums have been considered. Conventional, envelope-based bus compressors with stereo inputs and outputs often operate on the principle that the maximum of the detector signals from the left and right input channels is used as a single, combined detector signal, and both channels always undergo the same compression. This results in no shifts in the stereo image, and the currently louder channel, i.e., left or right, determines the compression behavior. If, as before, a virtual mono sum of the multi-channel compressor controls a number of input channels, and these are then panned and summed to a stereo signal, the result, with the same compressor settings, is generally different than if the stereo sum had been subjected to the described bus compression only after summing.The imaging function of the panorama control also comes into play here. The differing control behavior may not pose a problem in many cases, especially since no shifts in the stereo image occur in the case of the mono sum. However, if one wants to achieve exactly the same control behavior as in the case of the bus compressor, the following variant is recommended. Fig. 13 again shows a simple mixing console system into which a multi-channel compressor 200 is integrated. Mono channel strips 180-m are panned in a panorama unit 1600 and mixed directly to a stereo master bus 1700 via summing mixers 1710 and 1720. As shown, the sends 1510-m to effects units 550 are routed after compression, i.e., also to an output fader 190-m. The total positions of the panorama controls 1610-m in this arrangement are transmitted as information to the multi-channel compressor, for example, as panning information 1620.The detector path also contains control channel panorama controls 346-m, which are set using the panning information 1620 in the same way as the panorama controls in the panorama unit 1600 in the master bus 1700. The mono control channel signals 360-m are thus converted into stereo signals 361-m, 362-m, which are mixed to form a virtual stereo sum 371, 372. The detector unit 380 therefore has two detectors for determining a right and a left sum level and, if necessary, two detectors for each control channel to determine a right and left channel level. If all weights have a gain of 1 and the filters are also set to neutral, the virtual sums in the detector path of the compression control device are equal to the respective summed proportions of the channels concerned, which are summarized in the mast bus.In this case, a situation has been created where the multi-channel compressor accesses the same signal as a hypothetical bus compressor accesses the sum of the input signals, assuming only these channels are mixed onto the bus. If an identical sidechain filter is used for all channels, this correspondence would still exist if the hypothetical bus compressor used the same sidechain filter, but only for its two input channels (L / R) instead of the entire set of input channels. Only changing the weights results in different behavior, which may be desirable. Therefore, by using the copied panning in the detector path, it is possible to emulate a bus compressor using the multi-channel compressor. For the sake of simplicity, the following description is formulated only for global group processing.Partitioning for multiple parallel group processing operations proceeds in the same way as before. Panning a mono input signal is simply a dual mapping of the signal to two output channels in the form of an angle-dependent weighting. The weighting functions for an angle θ are given for the left and right channels by P. L (θ) and P R (θ). θ m Let the panoramic angle for the m-th input channel be . Then, from the filtered and weighted input signals according to Eq. (2), the following stereo tuples (c) result (c). L m (n),c R m (n)) through panning: ^ ^(^) = ^^ ^ (^) ∙ ^^(^^) (2.1)Applicant: STAGETEC GmbH 12.09.2025Our reference: P18.918WO^ ^ ^ ^ (^) = ^ (^) ∙ ^^(^^) (2.2)The virtual signal sum is this time a stereo tuple (sL(n),sR(n)) and replaces Eq. (3): ^^(^) = ∑ ^ ^ ^^ ^^ ^ (^) (3.1b) ^^(^) = ∑ ^ ^ ^^ ^^ ^(^) (3.2b) The individual components sL(n), sR(n) of the stereo sum and the individual components cL m (n), cR m (n) of the acquired stereo channels now pass separately through corresponding envelope detectors. Subsequently, the maximum of the left and right components is determined for each channel and the sum: ^^(^) = max {^ ^^^{^^ (^), {^ ^^^{^^ (^)}} (8b) ^^(^) = max {^^{^^(^)}, ^^{^^(^)}} (9b) The combined detector signals thus obtained d m(n) and se(n) for the stereo case serve as the basis for the subsequent process steps of the procedure as before and, in this role, replace the original detector signals in Eqs. (8) and (9). The stereo procedure is limited to group compression with envelope generators with manual or automatic thresholds. The methodology is less suitable for use with an automixer, and RMS detection is also omitted. Up to this point, stereo channels have been obtained through panning. In the case of true stereo input channels, these can be fed directly to the detector by simply mixing the left input channel to the left sum and the right channel to the right sum. The described multi-channel compressor based on the principle of a virtual sidechain bus can be considered a separate category.This compressor operates on the principle that while channels are compressed together using an internal sum and thus influence each other, they remain as individual signals and can be further processed. This makes the multi-channel compressor an alternative to a conventional bus compressor. Applicant: STAGETEC GmbH, September 12, 2025. Our reference: P18.918WO. If such a compressor is integrated into a mixing console setup or inserted externally as a standalone device via post-fader insert paths into all relevant channels, the classic problem of varying wet / dry mixes with integrated effects units can be solved without having to integrate these effects units into a common bus. In this case, the sends are located post-fader and after the compressor. Furthermore, the ability to weight channels and set individual ratio values ​​allows for processing techniques that are not possible with a conventional bus compressor.The presented compression with automatic threshold is also a novelty and is only possible because, as in this case, the compressor has access to all individual channels. The threshold is derived from the level of the loudest channel plus a constant offset. This mode can be seen as complementary to an automixer in a certain sense, because a channel with a higher level pulls the threshold upwards, thus resulting in less compression for the other channels, whereas an automixer does the opposite by significantly attenuating lower-level channels. This method can be applied to groups of singers or instruments. Thanks to the automatic threshold, the user no longer needs to set or manually adjust it. They only need to adjust the offset and ratio appropriately, and whether a single global setting is sufficient in practice is another matter.The multi-channel compressor operates on a mono sum, even if the relevant channels are later distributed in the stereo panorama and mixed to a bus. However, to achieve the same control behavior as a peak-based bus compressor, which uses the maximum signal levels of the left and right channels in the detector path, another variant was introduced. In this variant, based on provided panning information, the multi-channel compressor copies the resulting stereo sum of a master channel, thus exhibiting essentially the same control behavior as a bus compressor on the aforementioned master channel. This assumes that the multi-channel compressor is integrated as a block into a simple mixing console system where channels are always summed to a stereo master bus—a fairly common configuration. Applicant: STAGETEC GmbH, September 12, 2025. Our reference: P18.In section 918WOE, a sequential implementation was described in which linked neighboring channels can be spontaneously combined into automixes and group compressions and displayed as such, without having to create separate buses. Separate views on an integrated or connected screen or display are required for compression settings. The restriction to neighboring channels can be mitigated if the channel strips have source selection, i.e., an input router exists in the system. With the described two-step procedure for sum and maximum, the overall computational effort remains approximately constant, regardless of how the groups are partitioned.The increased effort required for all calculations related to the channel sum is justified by the fact that, with this implementation, the multi-channel compressor can perform up to M / 2 non-overlapping group compressions or up to M individual channel compressions when needed, assuming an even number of M input channels. The same procedure, as described, can also be implemented using parallel basic blocks that have the same controls for grouping adjacent channels and can even be implemented with analog signal processing. This is made possible by a forward and reverse sum that is passed between adjacent modules. Overall, the multifunctional multi-channel compressor with a virtual bus compressor offers...a dynamic processor is available in the audio channel control unit, which is equally capable of automixing and compression of individual signals and groups, and provides a powerful, flexible tool for audio editing.

[0002] Applicant: STAGETEC GmbH 12.09.2025 Our Ref: P18.918WO Reference Mark 1 Signal Processing Unit 11, 12, 13 Channels 20 Summer 30 Bus Signal 41, 42, 43 Send Channel 51, 52, 53 Weighting Unit 60 Summer 70 Sum Signal 80 Effects Unit 90 Effect Signal 100 Bus Compressor 110 Bus Signal 120 Additional Summer 130 Output Signal 180, 180-1, 180-2, 180-M Channel Strips 190, 190-1, 190-2, … 190-M Output Fader 200 Multi-Channel Compressor 205, 205-1, …, 205-M Control Knob 206, 206-1, …, 206-M Other Knobs / Link Buttons 210, 210-1, 210-2, …, 210-M Audio compression channels 211, 212, 215 Individual tracks 220, 220-1, 220-2, … 220-M Input 221 Link input 240, 240-1, 240-2, …, 240-M Output amplifier 250, 250-1, 250-2, … 250-M Output 260, 260-1, 260-1, …, 260-M Output signals 270 Bus signal 280 Bus summer 290 Master bus summer 300 Compression control device 310, 310-1, 310-2, …, 310-M Audio control channel 320, 320-1, 320-2, …, 320-M Input 330, 330-1, 330-2, …, 330-M Side channel filter Applicant: STAGETEC GmbH 12.09.2025 Our reference: P18.918WO 340, 340-1, 340-2, …, 340-M Weighting Unit 341 Detector Sections 342 Line Component Summer 343 Logarithmic Section 345 Maximum Finder 346-m Control Channel Pan Control 350 Channel Gain Calculation Unit 355 Summer 360 Audio Control Channel Signals 361-m, 362-m Stereo Signals 370 Sum Control Signal 371, 372 Stereo Sums 380 Detector Unit 390 Target Gain Signal 510, 510-1, 510-2, …, 510-M Send Mixes 520, 520-1, 520-2, …, 520-M Weighting Units 550 Effects Unit 560 Output Signal 710-1 Group compression 810-1, … 810-M incoming signals 820 Router 830 Mixing console 840 Multi-channel compressor (200!) 910 Input data 920 Output data 930 Processor 940 Memory 1100 Basic elements 1103-m Link information 1104-m Link switch 1110-m Channel signal 1120 Block 1130 Summer 1140-m, 1140-m-1 Forward sum m 1141-m Forward sum input Applicant: STAGETEC GmbH 12.09.2025 Our reference: P18.918WO1142-m Forward summing output 1145-m Switch 1150 Reverse summing 1151-m Reverse summing input 1152-m Reverse summing output 1155-m Additional switches 1160-m Detector 1170-m Logarithmic (summing) level 1180-m Summer 1190-m Summer 1195-m Compression channel detector 1200 Potential rail 1201-m Potential rail input 1202-m Potential rail output 1205-m Another switch 1210-m Channel level 1218 Maximum channel level 1220 Adaptive threshold 1240-m Selection circuit 1250 Channel gain calculation device 1258-m Target gain signal 1260-m Threshold 1270-m Influence factor 1280-m Constant threshold 1290-m Offset 1340-m Lower barrier sum 1350-m Total sum sm 1510-m Sends 1600 Panoramic unit 1610-m Panoramic controller 1620 Panning information 1700 Master bus 1701, 1702 Master bus stereo signals.

Claims

Applicant: STAGETEC GmbH 12.09.2025 Our reference: P18.918WO Patent claims 1. Multifunctional multi-channel compressor comprising a plurality M of audio compression channels (AKK-m), wherein each audio compression channel (AKK-m) includes an input (Em) and an output (Am) and a controllable output amplifier (compressor VCA-m) connected therein via an audio signal, and a compression control device (KSE) comprising for each of the plurality M of audio compression channels (AKK-m) an audio control channel (ASK-m) whose audio control channel input is connected to the input of the audio compression channel (AKK-m), and an audio control channel output (StA-m) which is connected to a control input (StE) of the controllable output amplifier (VCA-m), wherein each of the audio compression channels (AKK-m) is assigned a linking device for linking to at least one other of the audio compression channels (AKK-m),wherein compression mode selection device (KMW-m) is configured to specify a compression mode for each audio compression channel (AKK-m) or group of linked audio compression channels (AKK-m), wherein the compression control device (KSE) additionally includes at least one summing device (SE) for forming audio control channel summation signals (s, p (n)) (AVS-p) of interconnected audio control channel signals (c m (n)) the majority of the audio control channels (AKK-m) and at least one detector device for determining channel levels (D m (n)) for the individual audio control channel signals and sum level (Sp(n)) for the audio control channel sum signals (sp(n)), wherein the compression control device (CCD) has a channel gain calculation device (CCD-m) which calculates a target gain signal value for each audio compression channel as a function of the sum level S p(n), into which the audio control channel signal has been received, and a comparison threshold is calculated, wherein the comparison threshold is selected depending on the compression mode set by the compression mode selector (KMW-m) for the corresponding audio compression channel (AKK-m), wherein in automix mode the comparison threshold is the channel level (Dm(n)) and in group compression mode a fixed threshold constant (C p) is. Applicant: STAGETEC GmbH 12.09.2025 Our reference: P18.918WO2. Multifunctional multi-channel compressor according to claim 1, characterized in that each of the audio control channels comprises a side-chain filter (SFC).

3. Multifunctional multi-channel compressor according to claim 1 or 2, characterized in that each of the audio control channels comprises a weighting device (Wm).

4. Multifunctional multi-channel compressor according to any one of the preceding claims, characterized in that the signals representing the target amplifier values ​​determined in the channel gain calculation device (KVBE-m) are passed through a smoothing device before being used to control the controllable output amplifiers.

5. Multifunctional multi-channel compressor according to any one of the preceding claims, characterized in that the determination of the values ​​is carried out in a multi-stage iterative or multi-stage parallel process.wherein in an iteration one or more of the magnitude and / or one or more intermediate results for one of the audio compression channels (AKK-m) are determined, wherein in an iteration different magnitudes or different intermediate results for different audio compression channels (AKK-M) can be determined simultaneously, wherein the iterations of a stage are executed in a number corresponding to a number of the plurality M of audio compression channels (AKK-m).

6. Multifunctional multi-channel compressor according to one of the preceding claims, characterized in that it is operable in a panning mode in which the audio control channel distributes the optionally weighted and optionally filtered audio control channel signals according to panning factors that specify a proportional distribution for the respective channel across the panning tracks.split into N-tuples and, based on these tuple values, separately determine an N-tuple panning track sum level and an N-tuple panning track level, and determine the maximum panning track sum level as the sum level and the maximum panning track level as the channel level. Applicant: STAGETEC GmbH 12.09.2025 Our reference: P18.918WO7. Multifunctional multi-channel compressor according to one of the preceding claims, characterized in that the audio compression channels (AKK-m) are indexed and, via the linking device, the respective audio compression channel (AKK-m) can only be linked to an audio compression channel (AKK-m-1) adjacent with respect to indexing, or can be identified as not linked.wherein all audio compression channels (AKK-m) can be linked either to an audio compression channel (AKK-m) with a smaller index or all only to an audio compression channel (AKK-m) with a larger index.

8. Multifunctional multi-channel compressor according to one of the preceding claims, characterized in that, in addition to or as an alternative to the automix mode and / or the group compression mode, a group compression mode with automatic threshold can be selected, in which the automatic threshold is the current maximum channel level of the audio compression channels linked together to form a group plus a positive offset, and the automatic threshold is used as a threshold for calculating the target gain value signals.

9. Multifunctional multi-channel compressor according to claim 8, characterized in that an upper and / or a lower threshold for the automatic threshold can be set.by which the automatic threshold determined on the basis of the channel levels and the positive offset is limited downwards by the lower threshold and / or upwards by the upper threshold.

10. Multi-channel compression method for a plurality M of audio signals, comprising the steps of: feeding audio signals into a plurality M of audio compression channels, each of the audio compression channels having an input and an output and a controllable output amplifier arranged between them; establishing links between the individual audio compression channels to form groups of channels; determining the individual gains for the output amplifiers of the individual audio compression channels by side-channel processing from the audio signals of Applicant: STAGETEC GmbH 12.09.2025 Our reference: P18.918WO audio compression channels that are grouped together,a sum signal is generated for each channel, and a sum level is determined based on the sum signal. The control signals representing the gains for the individual audio compression channels are determined by comparing the sum level, into which its audio signal has been incorporated, with a threshold value. An attenuation of the gain occurs when the sum level exceeds the threshold value. For each channel group, a compression mode is selected from a set comprising a group compression mode and an automix mode. When the automix mode is selected for a group of linked audio compression channels, a channel level is determined for each of their audio signals, and the channel level is used as a threshold value in determining the audio compression channel-specific gain. A compression constant is set for a group and used as a threshold value in determining the audio compression channel-specific gain.when the compression mode "Group compression" is selected for the group.

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