The present invention pertains to audio processing units. More particularly, the present invention pertains to audio processing units which can be incorporated into an audio processing unit. Even more particularly, the present invention pertains to audio processing units for processing an inputted audio signal via an audio processing unit.
A rack frame is a standardized frame or enclosure used for arranging multiple electronic equipment modules usually in a vertically mounted fashion. Modules or “units” to be mounted within the rack frame are typically 19 inches wide and include an “ear” on each side of the unit to facilitate mounting the unit to the frame with the use of any suitable fasteners. These frames are typically found in computer server rooms, audio recording studios, scientific labs, and the like. As such, common units include, firewalls, switches, audio processors, and the like. The height of a units is measured in “rack units,” abbreviated as RU or U. One RU is equal to 1.75 inches. The height of the unit is directly related to the amount of equipment stored within the unit. A unit may have a width less than 19 inches and not be mounted to both sides of the rack. However, the unit will typically require a larger height in order to adequately support the weight of the unit.
Audio processing units generally include any number of inputs and outputs for receiving a raw audio signal and transmitting a processed audio signal, respectively. In adjusting the signal received by the audio processing unit, the unit includes a plurality of controls for adjusting different levels of volume and frequencies and control equalizers, etc. However, the processing capabilities of the audio processing unit is directly limited by the digital signal processor or DSP within the unit itself. As a result, many commercially available racks for processing an audio signal are limited in their ability to master or remaster an inputted audio file.
Therefore, there is an ongoing need for an audio signal processing unit which provides improved audio mastering capabilities and allows for user controlled customization of the mastered audio signal.
It is to this to which the present invention is directed.
The present invention provides an audio processing unit comprising a sound enhancement microchip having software embedded therein. Preferably, the audio processing unit is incorporated into a frame rack.
The unit includes at least one input for receiving an originally unprocessed audio signal and at least one output for transmitting the resulting processed audio signal to a source for review and/or storage.
The software embedded in the microchip improves the quality of the audio signal being sent to the unit prior to any user modification. The microchip operates by duplicating the inputted audio signal, splitting the duplicated signal into at least two exact duplicates of the original inputted signal, layering the signals, processing the layers, and recombining the layers to provide an enhanced or processed audio signal.
The circuitry associated with the microchip can be provided as a single integrated circuit or a microcircuit. All processing is done in the digital domain.
The microchip is interposed within the existing circuitry of the unit, at a convenient point, prior to the delivery of the audio signal to the user.
Thereafter, the processed audio signal is available for further user processing by utilizing any of the controls on the unit.
For a better understanding of the present invention, reference is made to the accompanying drawing and detailed description. In the drawing, like reference characters refer to like parts through the several views, in which:
The present invention is directed to an audio processing unit including a sound enhancing microchip disposed therein. It is to be understood that the audio processing unit hereof has particular utility in professional recording studios, broadcast facilities, and other audio processing environments.
Now, and with reference to
It is to be understood that the audio processing unit 10 comprises an enclosed housing 14 having a front surface 16, a rear surface 18, a top surface 20, a bottom surface 22, and a pair of sides 24, 26. Preferably, the housing 14 is dimensioned to fit within a standard open-frame rack (not shown). Thus, the housing 14 includes a plurality of apertures 28 formed in the corners of the front surface 16 of the housing 14 for allowing suitable fasteners (not shown), such as screws or the like, to mount the unit 10 to the rack.
The unit 10 may include a plurality of knobs, buttons, meters, plugins, and other suitable controls for allowing user adjustment of a processed audio signal.
As shown in
Referring, again, to
Preferably, the suitable sample rates that work with the unit 10 are 32 k, 44.1 k, 48 k, 88.2 k, and 96 k. Alternatively, these sample rates can be selected in combination with the equalizer settings to provide the desired audio output.
The housing 14 also includes an RMS meter 50. The RMS meter 50 shows the input and output levels of both left and right channels. A meter (input) button 52 is included for selecting between two metering options. Additionally, a bypass button 54 is included on the housing 14 which will output the original, unprocessed audio source. In the case of a power failure or when the unit 10 is powered off, all digital and analog input and outputs will automatically be bypassed. Such RMS meters are well known and commercially available.
It is to be understood that any other controls other than those specifically described above may be included on the housing for allowing various modifications and/or adjustments to be made to the processed audio signal.
The unit 10 also includes a plurality of different inputs and outputs for receiving and transmitting the audio signal to a variety of acceptable audio sources. As shown in
The unit 10 also includes internet connection capability via an ethernet cable (not shown). As such, the unit 10 includes an ethernet input 68. This allows the settings on the unit 10 to be changed remotely when connected to the internet. Alternatively, permissions on the unit 10 may be restricted such that the settings on the unit 10 can only be modified by designated persons via the internet connection.
Preferably, the unit 10 includes an external power supply (not shown) as well as the ability to run redundant power via the XLR input 56 or, alternatively, an AWG universal power input 70.
As noted above and in accordance herewith, the unit 10 has a microchip 12 included therewith. The chip 12 is a digital signal processor (DSP) for processing the inputted audio signal prior to further processing by the user. The microchip 12 is directly connected to the existing circuitry of the unit 10 and operates to master or remaster the inputted audio signal once received by the unit 10.
It is to be understood that the circuity of the unit 10 and the microchip 12 are each in electrical communication with one another by any suitable wiring and configuration well known to one of ordinary skill in the art.
Referring now to
According to the present invention, generally, sound enhancement is achieved by the microchip 12 receiving the audio signal from a circuit board 72 and creating both an exact duplicate of that initial signal and at least one secondary signal. The at least one secondary signal is an exact duplicate of the initial signal, thereby creating at least two layers therefrom. Optimally, each of the layers is processed within the microchip 12 by an equalizer calibrated at a selected frequency. The frequencies of each layer can either be the same or different from each other.
At least one of the layers, preferably at least two layers, is processed. Once the at least one layer is processed, the signals are layered and passed through an equalizer, their volumes adjusted, passed through a leveler and, lastly, a master fader of the layered signals are being combined via a combining bus, as further discussed below.
As shown in
Initially, an equalizer 222 either filters out any unwanted frequencies or boosts or adds frequencies in the duplicate. As noted above, the output from the equalizer 222 is split into at least one copy of the initial signal and one secondary signal, which is a duplicate of the signal of the initial signal. The signals are then processed and layered. This is shown as Layer 1, denoted at 223, and Layer 2, denoted at 233. Frequencies in the range of 125 to 400 cycles per second can be adjusted to any desired levels or volume, any subsequent layers will be affected by this reduction. The use of the equalizer 222 can, if desired, be eliminated, but has found to be important in facilitating processing of the signal during subsequent processing of each of the layers 223, 233.
As described below, if desired, more than two layers can be produced such as Layer 3, Layer 4, etc. which can be processed in the same manner shown in
During the processing of Layer 1223, Layer 1233 is first adjusted in volume by fader 223a; processed by an equalizer 224; adjusted in volume at 225 via an equalizer volume control; compressed by compressor 226; adjusted in volume by a compressor volume control 227; processed by compressor/expander 228; has its volume once again adjusted by the fader at 229; combined with the secondary layer 260; processed by equalizer 230; adjusted in volume, again, by an equalizer volume control at 230a; and, lastly, processed by limiter 241 where the layered signal is sent to the microchip output 38 through master fader 241a and output 241b.
Layer 2233 is processed in the same manner as Layer 1223 and, therefore, Layer 2 is first adjusted in volume by fader 233a; processed by equalizer 234; adjusted in volume by the equalizer volume control at 235; compressed by compressor 236; adjusted in volume by the compressor volume control at 237; processed by compressor/expander 238; adjusted in volume again at the fader 239; merged atop or layered atop Layer 1 at a combining bus layered signal 260 prior to entry into the equalizer 230 where the merged layers are processed by equalizer 230; undergoes final adjustment in volume 230a by the equalizer volume control; processed by limiter 241 and outputted to the microchip output 38 through master fader 241a and output at 241b.
More particularly, just prior to entry into the equalizer 230, the combining bus 260 is used to layer the incoming signals, followed by the layered signals being processed together. The output 241b is thus the layered enhanced audio to be sent to the handset 26.
Synchronization of the processing of all layers is important. The time required for each layer 223, 233 to pass through its respective processing is substantially equivalent so that the signal of each layer 223, 233 takes substantially the same amount of time to pass through its processing, merge at the combining bus 260, and be outputted as a single layered audio signal at 241b. Optimally, processing is done at the same time.
The signals produced by each layer 223, 233 can be equal in loudness, but in most cases, usually Layer 1223 is louder than Layer 2233. For example, Layer 1223 can have its bass minimized while emphasizing and processing higher frequencies. Similarly, Layer 2233 can have its higher frequencies minimized while emphasizing bass frequencies or vice versa.
If the bass frequencies in the initial audio signal are weak, processing of Layer 2233 can increase the loudness of the bass frequencies so that when the processed layers 223, 233 are joined at the combining bus 260, prior to steps 230, 230a, 241, 241a, and 241b, the resulting audio signal increases a bass component with a greater volume and presence than is the case in the initial audio signal or vice versa.
Layer 1223 and Layer 2233, each, ordinarily, focuses on a band of frequencies that is different from any band of frequencies focused in the other layer. The frequencies that are not being focused on in one layer are being focused on in another layer and complement each other.
After enhancement, the dynamic range appears to be retained.
When a compressor, such as at 226, 236, is utilized, the threshold setting is typically adjusted to the user's preference. Preferably, each layer 223, 233 is processed with the equivalent of at least one piece of enhancement equipment simultaneously. While the processing shown in
In general, the particular frequencies that Layer 1223 or Layer 2233 emphasizes will experience an increase in volume compared to their volume levels when the frequencies first enter the onset of the layering process from the equalizer 222. The signal processing is akin to that disclosed in the aforementioned co-pending U.S. Patent Application.
The sound enhancement may be accomplished through portable circuitry such as on a pc board, or may be incorporated into a microchip such as disclosed in the aforementioned co-pending U.S. Patent Application or in a software app.
As disclosed in the co-pending application and with reference to
In the embodiment represented by
Referring now to
The contents of I/O LUT 512 may vary depending upon which layer N is being represented by circuitry 500. The compressed output of gain level control or multiplier 508 is then applied to a compressor/expander 514, wherein the same compressed output is applied as an input after splitting to both gain level control or multiplier 516 and band level detector 518. The output of band level detector 518 is applied to a gain LUT 520, which provides a gain control input for gain level control or multiplier 516. The circuitry of band level detector 518 and/or gain LUT 520 may vary depending upon which layer N is represented by circuitry 500.
The output of compressor/expander 514 is provided as an input to gain level control or multiplier 522, which, together with applied signal GAINLN_2, acts as an overall gain control for Layer N.
In the embodiment represented by
Similarly, in the embodiment represented by
Likewise, in the embodiment represented by
Further, in the embodiment represented by
After the sound enhancement processing is completed within the circuit 400 of the microchip 12, the layered audio signal is transmitted through the remaining circuitry of the unit 10 in order to be further processed by the user by utilizing any one or a combination of the knobs and buttons on the housing 14.
While the unit 10 disclosed hereinabove is described as being mountable on a frame rack, it is to be understood that the unit 10 may be situated on any surface without deviating from the scope of the present invention.
From the above, it is to be appreciated that defined herein is a new and unique mountable audio processing unit having an audio enhancing microchip disposed therein for processing an audio signal and improving the quality of the audio signal prior to being delivered to the user for further processing.
This application is a continuation-in-part application of co-pending U.S. patent application Ser. No. 15/091,595, filed Apr. 6, 2016, for “Microchip for Audio Enhancement Processing,” which claims the priority benefit of now expired U.S. Provisional Patent Application Ser. No. 62/143,253, filed Apr. 6, 2015, for “Microchip for Audio Enhancement Processing,” the entire disclosures of which are hereby incorporated by reference in their entirety including the drawings.
Number | Date | Country | |
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62143253 | Apr 2015 | US |
Number | Date | Country | |
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Parent | 15091595 | Apr 2016 | US |
Child | 16114378 | US |