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Manley Neo-Classic 300B RC Preamplifier with Remote Control The 300B Preamplifier is Manley's most obscure product. The Neo-Classic 300B Preamplifier RC was handcrafted in Chino, California, and has great-sounding components in a creative and clear signal routing. So it is the first and only time these vintage tubes have been used in a preamp. For many years, Manley has been using the 300B into modest power amplifiers, and the admirable thing about the 300B is the tube's smooth, natural tone. A single 300B, on the other hand, offers genuinely suitable power and voltage swing for headphones. It also serves as a fantastic line driver. This is where the impedances aren't too low, and the power needs aren't too high, which is excellent for a single-ended tube of this type. For the input stage, Manley blends the vintage 300B with another classic, the massive octal-based 6SL7GT-all triodes operating in true Class A mode. Manley has used vacuum tube rectification and control in the power supply. They have added much capacity in order to supplement the vintage vibe. Manley also added an extra and unique ripple rejection circuit to remove the noise in the power supply, resulting in a very low-noise preamplifier, which is very important for people using headphones. The Neo-Classic 300B preamplifier blends the legendary 300B tube with the big octal-based Sovtek 6SL7GT for the input stage. Every triodes runs in pure Class A mode. To cut the noise in the power supply, Ruby 5AR4C tube rectification, Raytheon JAN OD3 tube regulation, and a ripple rejection circuit is used. The input impedance is lesser than the Jumbo Shrimp when kept at 100k Ohm. But the output impedance is greater than the Jumbo Shrimp at 100 Ohms. The Neo-Classic 300B is not only a headphone amplifier but also a line-level preamp. The headphone part is very impressive and is powered by an identical pair of 300Bs. There are two headphone jack inputs and a toggle switch on top of the machine that allows you to choose between a 30-400 Ohms or a 300-4000 Ohms headphone impedance output. It has a very wide, clear sound that is simple to listen and the sound is incredibly accurate and realistic, which may be described as organic. Features of the Manley Neo-Classic RC Preamplifier with Remote Control INPUT SELECT- The Input selection is switched to 5 positions. Electrically, each input is identical. This switch is immediately followed by the RECORD OUT. If the recording is not done, the interconnect cables must be unplugged from the RECORD OUT since there is no buffer amp to separate the effects of loading and cable capacitance. HEADPHONE JACKS - The headphone jacks are standard1/4" stereo. The HEADPHONE IMPEDANCE SWITCH should be set to the appropriate range for the headphones. The real headphone impedance determines the maximum power output. The preamp will convert 1 watt (10 volts RMS) into 100 ohms and 1 watt (25 volts RMS) into 600 ohms25 volts RMS) into 600 ohms as the output. These outputs are transformer linked for maximum protection from current shock, shortage of cable, and other potential issues. VOLUME - The volume is featured with an Audiophile-grade motorized ALPS potentiometer. The remote can be used to control the volume. BACK-LIT PANEL -The BACK-LIT PANELindicated the power. The unit is turned on when it is lighted up, and it is off if it is not lightened. POWER SWITCH - Since the tubes take a while to warm up, firstly around 10 seconds, there is stillness, and then there is a different sound produced, after which it becomes normal. It is better to switch on the CD to power up the system and then turn on the preamp and finally the amplifiers for good sound. After waiting for a minute or two, start playing music. Everything will warm up as a result of following this process. After an hour or two, some individuals might experience a difference, depending on the equipment. When powering down, the order is changed: the power amps come first. This is a wonderful habit since it puts the least amount of strain on the speakers and ears. As the preamp's power supply is tube-based, this procedure would benefit the slow start to speed up. Switch on On the right side of the front panel, you'll find the power knob. The preamp may be turned on or off by turning the knob to the right or left. It is not advisable to turn the knob left or right in a rapid motion. A lengthy garolite 1/4" rod shaft links the faceplate knob to the real switch in the rear of the unit. This is present in the knob and on the input selector. The small screws on the shaft-to-switch coupler will become loose if the knob starts spinning around in circles. If this arises, the unit must be unplugged, then remove the bottom cover, and use a 1/6" hex key to tighten the four coupler screws. Thus, the issue gets solved. Running It's not is not ideal to leave your preamplifier turned on all the time. This is a waste of power as well as tube life. The preamplifier features both tube and solid-state rectification, and it takes around 30 minutes to achieve peak functioning condition. Like how running is beneficial to the musculoskeletal and cardiovascular health, there is a certain period of time that the preamplifier should run to be in good condition. Tube Life The quality of these tubes, like all tubes, declines over time. This is due to cathode emission, which occurs naturally in all tubes. Depending on usage, it is good to have your preamplifier tested every 4-5 years. The need to replace the 6SL7 input tubes might be indicated by an extreme rise in noise level.  Generally, the 300B's are known to endure an extremely long period.  Headphones There is a switch on the top of the preamp, and you may select the right impedance for your headphones. When using headphones, it is recommended to pick the "Transformer" output rather than the "Direct" output because the transformers are favorite headphones over the direct output. Features of the Remote Remora Control For optimal signal selectivity, flexibility, and system range, the remote transmitter and receiver are controlled by a crystal-clocked microcontroller.  The user can reconfigure the user-number component of the serial ID data stream from the distant system. This eliminates the risk of unintended behavior caused by remote command clashes.  When the transmitter is in operation, a multi-color LED indicates battery status and two-digit user-number programming status when it is in program mode.  Internal antennas assist in a streamlined design look and prevent the risk of harm if exposed to the elements while still offering enough range.  The transmitter is powered by a 9V battery that is widely available.  For durability, the transmitter shell is composed of strong Lexan®.  For years of consistent operation, real compression-dome push buttons are employed rather than short-lived conductive silicone membrane pads.  For the essential audio channel, a high-quality motorized attenuator is used, with excellent inter-channel matching.  When the button is held for half a second, the attenuator motor starts to run at a slower pace, and after a few seconds, it rotates at double speed. The keep-alive period on the remote transmitter keeps the Remora Remote awake for a few seconds AFTER either button is released, as shown by the LED bulb flickering DIM green or red.  The backup battery is unnecessary as the transmitter and receiver-CPUs decoder includes flash memory.   As a result of the clever system architecture and user-programming method, dip-switches are no longer required.  The Remora Remote uses relatively little power, resulting in long battery life.  The preamplifier's audio route has less noise because of its electrically quiet operation. 

An introduction to the Manley SNAPPER Power Amplifier The Snapper got introduced in 2002, a very delectable all-tube 100-watt monoblock. Mitch Margolis' Snapper utilizes a quartet of EL34 or 6CA7 transistors in a completely differential circuit layout, including the input and driving stages. The XLR input on Manley Snapper is driven fully balanced. If the preamplifier has balanced outputs, this amplifier is the first Manley design to utilize your XLRs without the use of extra balancing input circuitry or input transformers. The unbalanced RCA's can be used instead of balanced RCA's in this particular topology with similarly excellent results. Manley Labs' monoblock amplifier is as unique as the company itself. This company, which is owned and run by a woman, is an outlier in an industry. The company workers are mostly men whose hardworking reap a great production and success in the majority. Manley Labs is owned by EveAnna Manley. Like other designers and manufacturing engineers, she is also one among them. The main purpose of Manley is to make your recordings sound professional. Many A-list recording studios throughout the globe use their tube-based microphone preamps, limiters, equalizers, line preamps, D/A converters, and power amplifiers. For Manley, Home theatre is neither the first nor second interest, Manley, like Genelec, M&K, Bryston, and ATC- all of whom specialize in professional recording gear—does, however, provide specific components that can utilize in a high-end home cinema. This is the case with their new Snapper monoblock amplifier. The Snapper features four EL34 power tubes, each of which generates a certain amount of heat, although the amp's structure allows for adequate ventilation. Even though the Snapper, like other tube amplifiers, requires periodic biassing of its power tubes, the technique is so easy that it can be done in minutes.   Although it's rectangular, two of its front corners have been sheared off to create a shortened plane on the front face. A lighted, 114 by 212-inch panel reads Manley snapper on the metallic-blue faceplate. A 1-inch-thick metallic-blue dowel serves as a spiked foot on each of the Snapper's corners, lifting the unit's bottom plate by nearly 112 inches. The dowels not only help with ventilation but they also help to break up the cabinet's straight edges. Setting the upper part of the Snapper's back panel at a 45° angle to make the inputs and outputs easier to connect was another excellent touch. The Snapper is 39 pounds in weight. Six tubes make up the Snapper's basic yet complex circuit. The input tube is a 12AT7, with the voltage driver being a 7044. The output power is provided by four EL34 tubes working in ultra-linear mode. The complete amplifier is designed using a fully balanced differential topology, which is typical in solid-state amplifiers but uncommon in tube amplifiers. The negative speaker terminal is not the same as the positive output terminal. When compared to a traditional tube amplifier, this differential circuit enhances linearity. Manley Labs creates its own audio transformers, and the Snapper has a brand-new 19-section output transformer. Four 1300F, 350VDC computer-grade capacitors store 180 joules of energy for the B+ rail in the power supply. The Snapper monoblock from Manley Labs delivers the most incredible sonic features of tube-based power amplifiers into the home theatre at a price comparable to top-tier solid-state amps. Without compromising sweetness or slam, the Manley Snappers offer the finesse and midrange layering to reward deep absorption into complex soundscapes on the one hand and unadorned vocals on the other. The Snappers are a pair of monoblock tube amps that are both strong and quiet. They're sung in a way that sounds as beautiful and clear as rainfall while still packing a powerful dramatic impact. Together, these amps generate beautiful music. They're robust, tuneful, optimistic, and energetic, with just the proper amount of lushness and fluidity to create a sophisticated, emotionally engaging sound. Many subtle and overt modifications in the SNAPPER's circuitry enable uncolored and revealing melodic replication of the input signal. The all-balanced and self-balancing conveyance of this musical signal via the amplifier, from input jacks and speaker terminals, is paramount. This method helps to eliminate the undesired interaction between the audio signals passing through a particular amplifier stage and the power supply energy biassing that stage. The input and driver stages are rock-steady stabilized as a consequence of this method, and their performance is unaffected by signal dynamics or power supply voltage changes. Even though the signal is totally dependent on the existence of energy from the power supply, on which the signal rides, the two variables here are referred to as power supply and signal. Getting connected to the Manley Snapper First, you have to plug all the cables. Then you have to connect the IEC power cord to the Snapper. To prevent your Snapper from magnifying any of your upstream equipment's waking-up noises, switch on all of your upstream devices first: sources, preamplifiers, and so on. In the IEC power in let, you can find the power switch, which is at the back of the chassis. There is a Rocker switch next to the AC power cord. If you want to switch on the Snapper, then you have to switch away from the power cord. If you're going to Switch off the Snapper, then flip towards the power cord. Use of the SOFT-START Technology The SOFT-START technology on your Snappers is supposed to balance the rise or fall in the electrical circuit and any changes caused during power-up. When the power is switched on, the amplifier begins with a gentle warm-up procedure that lasts around 30 seconds. You should wait for the input and driver stage wake to be dissipated. Till then, the control of the output tube must be kept at the entire negative bias. During the warm-up phase, the badge lamps will blink, and the amp's inputs will be muted to prevent any signal drive until all stages are ready. You are not supposed to keep high-level line signals in the input. When the small input tube's heater kicks on, you can notice a flare of light. When tube heater filaments are cold (room temperature), they conduct a lot more current thus, they light a little brighter when they're first turned on until they reach normal working temperature. The tungsten filament in every incandescent light bulb accomplishes the same thing. That's very normal, so don't be concerned. Please remember that it takes around 45 minutes for the system to attain thermal equilibrium after hearing sounds from the Snappers for the first time. Power down in the Snapper As a general guideline, turn on your amplifiers last and turn them off first to avoid amplifying any stray sounds that may occur upstream from source components turning on or off. When the Snapper is not in use, it is also advisable to turn off the electricity rather than leave it on endlessly. This will extend the life of the tube and the system. Under normal circumstances, the tubes should last thousands of hours. Especially if you check your BIAS on a regular basis. It's with the continuous use of the Snapper that you will get to know how to operate it and will understand how the biassing should be done. Features Of the Snapper The signal path of the SNAPPER amplifier includes floating self-balancing input and driver circuits that are able to handle unbalanced RCA or higher-level balanced XLR signals. The input stage is designed to create a complementary push-pull signal with balance accuracy within a few percent of ideal when supplied with an unbalanced source. The following driver stage has the same self-balancing feature as the previous one but at a greater power level. This yields a finely tuned signal balance whose precision is unaffected by input signal symmetry and is limited only by the overall plate load matching accuracy of the driver [7044] tube.

An introduction to the Manley Stereo Limiter Compressor The Manley Nu Mu Stereo Limiter Compressor combines the iconic Variable Nu Mu T-Bar mod tube front end with the same ultra-smooth side chain and timing characteristics for a faster and punchier audio path, accompanied by a high voltage discrete and FET solid-state amplification and output stage for a quicker and punchier audio path. It's essentially a "clean" version of the Variable Mu, made for modern digital and electronic music when color and masking aren't as necessary as they were in the early days of digital. The Manley Nu Mu, which includes Manley's new HIP function, fine-tunes dynamic control to bring lower-level details without suffocating thrilling transients. The Manley Nu Mu is smartly developed to build efficiently and aesthetically to evoke gorgeous compliments from your clients, both visually and ergonomically. It's all proudly hand-built in California. 1. CONTROL OF THE OUTPUT. On both the L and R channels, the OUTPUT control is provided. It changes the volume of that audio channel's output. The ninth marking, or 11 o'clock, is when you reach unity. 2. THRESHOLD Control. On both the L and R channels, the THRESHOLD control is provided. THRESHOLD, also known as gain reduction, determines the required amplitude for compression or limiting to work. The most intense effect is when the control knob is set to MIN or totally counterclockwise. 3. LIMIT / COMPRESS SWITCH. When the switch is engaged and lighted, the Nu Mu can employ two compression ratios: compression (roughly 3:1) or limiting (about 10:1). 4. BYPASS / IN SWITCH The Nu Mu's BYPASS switch is a 'Hardwire' bypass for best performance. The audio will travel through either the Limiter or the Compressor if the switch is engaged and light to the "IN" position, allowing all of the front panel controls to operate. This setting ensures that the limiting or compressing isn't destroying the original sound but rather enhancing it or at the very least balancing it off. 5. VU METER. The Nu Mu's large bespoke VU meters simplify tracking both channels. When the switch (7) is set to REDUCTION, it displays the gain reduction in dB from the 0 dB point. The meter monitors the output in decibels when this switch is set to O/P and lighted. 6. HIP CONTROL. The "Hip" control compresses the overall signal at lower volumes (lower in the dynamic range spectrum) while leaving the louder dynamics alone. You get louder soft passages and the louder sections' natural dynamics. Parallel compression, in which the compressed signal is combined with the uncompressed signal, was previously the only option to get around this problem. 7. METER SELECT SWITCH. The Meter select switch allows the user to watch either the Gain Reduction or the Unit's Output. The output of the channel you're monitoring is seen in the associated VU Meter while the switch is engaged and lighted. 8. ATTACK CONTROL. The ATTACK control affects how long it takes the Compressor to respond to impulses that are higher than the threshold. The attack is a continuously variable control that affects the length of a transient required to trigger gain reduction. SLOW 0.13s, MID 0.07s, FAST 0.013s, all with a 63 percent gain reduction. SLOW 0.64s, MID 0.35s, and FAST 0.064s with a 90% gain reduction. Most percussive signals will not cause limiting or compression if you turn the knob completely counterclockwise at the lowest level. It can be found on both the left and right channels. 9. RECOVERY CONTROL. When the signal goes below the threshold level, the RECOVERY control specifies how long it takes the Compressor to recover. Five options with a 63 percent gain decrease recovery: 0.1s, 0.16, 0.32, 0.64s, and 1.7s. With a 90% gain decrease recovery, five options are available: 0.227s, 0.368s, 0.736s, 1.6s, and 3.87s. 10. LINK FUNCTION. When you utilize the LINK function, it connects the two stereo channels. The THRESHOLD, ATTACK, and RECOVERY CONTROLS are all linked through the LEFT CHANNEL CONTROLS. 11. INPUT LEVEL SWITCH. A toggle switch is used to regulate the input level. Each channel has its own Input Level control. There are three different levels of attenuation to choose from. +3dB, 0dB, and -3dB are the three decibel levels. 12. HP SC SWITCH. The High Pass Side Chain filter is set to 100Hz by default. When this switch is turned on and lighted, everything above 100Hz will be compressed or limited. If you want the bass to come through loud and clear while limiting or compressing everything else, this is the setting to use. 13. POWER SWITCH . When an IEC power lead is attached to the unit, the Power Switch emits a soft blue color. The Nu Mu is in STANDBY mode, as indicated by this. The Nu Mu is switched from standby to warm-up mode when the power switch is pressed. The backlit VU meters will begin to flash as the power switch illuminates to a brighter blue. The MUTE relays will lift, and the VU meters will stop flashing after this 30-second warm-up period. Manley Mic preamp Manley created the first modern 300B high-fidelity amplifiers, as well as the first modern high-powered tube power amplifiers and the KT90, the first new power tube design in thirty years. Manley was also the first in the United States to use ULTRA ANALOGTM converters, as well as the first in the world to demonstrate a converter with tubes. This website features the world's first 300B preamp as well as the first amps capable of switching topologies between single-ended and push-pull. From the early 2000s to the present, Manley has concentrated on reducing production methods and being more efficient while adhering to the brand's guiding values of high-end audio quality and American manufacturing integrity. The Tube mic preamps from Manley are a modern classic that gives unmistakable clarity and detail to every sound that goes through them. You can now enjoy the harmonically "alive" Class A sound in a simple UnisonTM-equipped plug-in, which is ideal for Apollo audio interface users. The MANLEY all-tube microphone preamplifiers come in 2-channel and single-channel configurations and are designed to work with modern capacitor (condenser) microphones. By adjusting the amount of total negative feedback on the GAIN switch, 40 to 60 dB of gain, enough for most ribbon mics, may now be selected in exact 5dB steps. This is a fascinating function, as it allows you to modify the sound's slew rate (speed), location, and tonal balance from mellow and "tubey-er" to faster, punchier, and more forceful. Both variants use the same tube complements one hand-selected 12AX7EH for low noise and one 6414 (or 12BH7) for each high-current output stage. Each component is hand-wired with silver solder and audiophile-grade components. Our current power supply design approach is demonstrated by double insulated custom-built power transformers and custom-built high-capacity reservoir capacitors. When you need to run your tunes via some tubes before hitting a sterile digital recorder, the front panel 1/4" DIRECT INPUT jack makes these preamplifiers double duty as a pure tube line stage for instruments including bass, drum machines, and keyboards. In this setting, the total gain is 20dB lower than the GAIN switch indicates position. Advantages of the Manley's Mic -preamp Manley's Class A tube mic preamp gives vocals and other instruments a smooth, clear sound. Using Apollo audio interfaces, record "through" the legendary Manley preamp in real-time. Make your tracks come alive by adding tube detail and complicated circuit behaviors. UnisonTM technology allows you to control Apollo mic preamp gain staging and impedance from the Manley Preamplifier plug-in. The Manley Tube Preamp plug-in, like the hardware, adds a smooth "HiFi" quality. Simply place it in the Unison slot of your Apollo interface's Console or LUNA software to bring voices and bass to life, punch up acoustic guitars and percussion, and bring synths forward.

An introduction to the Manley Stingray Stereo Integrated Amplifier The previous Manley 50 Watt Monoblocks were merged into a stereo chassis with a passive preamp, which resulted in the STINGRAY design. The final objective was to create a vacuum-tube integrated amplifier with absolute influence that upped the ante of high-end performance in a practical, accessible, bare-bones chassis with a dash of elegance. During a break for the cause at HI-FI '98 in Los Angeles, Eve Anna Manley had been sketching her hexagonal work-in-progress for former Stereophile Pooh-Bah J. Gordon Holt on a bar napkin, and that seemed to be a stingray. The history of making the Stingray emerged through this idea. The benefits of having a passive preamp are widely known: near-perfect clarity, no additional sound, and a signal route that is extremely simple. The Stingray's low level of global feedback, which has the effect of increasing input sensitivity, lends itself nicely to a passive front end.   The Manley Stingray is essentially a fixed-bias, dual-monoblock arrangement on a single stainless-steel chassis, with only one power transformer. It contains a four-position silver-contact input selector switch, screw-down-style binding posts, and separate left and right line inputs (gold-plated brass with Teflon dielectric) positioned to each side of the chassis and adjacent to their respective output transformers. Manley wanted to maintain signal lines and connections as minimal and direct as possible so he could use a passive preamp instead of using a gain-stage device therefore the Stingray's hexagonal shape, which is basically square-shaped with the front and rear corners were chopped off, and thus the hexagonal shape for Stingray emerged. As a result, a pair of bullet-shaped metal pillars hide and embellish the left and right corners, and it also serves as the amplifier's conical feet. The preamps had two primary functions such as switching and volume. A passive preamp would work great when all the sources are connected as a couple to the output. It is the same as in the Stingray. So it is better to set the passive preamp high to work well. But some people fall into a problem as they keep the passive preamp less. The amount of gain and dynamic range on the Stingray is the most impressive features. Manley and her design team were able to get four small EL84 output tubes per channel to work in ultra-linear mode.  So before striking the first 12AT7 input tubes, separate left, and suitable silver-contact select switches, which are there for the four stereo line inputs, will feed the music into the quality Noble balance and volume controls. The benefits of a passive preamp, which drives a two-inch cable with extremely low capacitance and near-absolute transparency, are well-known. They have no additional noise that would damage the quality of music, and the signal path is extremely simple. The dependable EL84 output stage may be adjusted between roughly 20 watts of TRIODE or 40 watts of Ultra Linear mode push-pull operation after the powerful 6414 driver/phase splitter. The trimpots and test points, which are conveniently placed on the top surface of the amplifier, may be used to modify individual bias for each tube. The power supply on the Stingray is extra-rugged and rigid, which is a MANLEY trademark. It is compared to with the sharks, which swim fastly, known for their agility, fluidity, and with who has complete authority. The TAPE LOOP, SUBWOOFER OUT, and RECORD OUT are all inactive. The controls for BALANCE and VOLUME are similarly passive. These circuits don't employ any active electronics or buffers. The RECORD OUT button is directly connected to the input choice switch. So, regardless of whether the Stingray is on or off, whatever source is chosen will be transmitted through the TAPE OUT. When the Stingray is switched off, the SUBWOOFER OUT will also create a signal, but you may turn down the volume control if you don't want any type of tones coming out of it. Front Panel Balance - This Noble balance control is unique and special. This balancing control has no loss in the middle and may be thought of as a straight wire position. No more than one channel's volume can be reduced. Volume– The volume is exceptionally a noble quality which has a conductive plastic precision attenuator Backlit Panel- Instead of incandescent bulbs, it employs ultra-bright LEDs for badge illumination. This panel indicates the power. If the IEC mains is connected properly and the On/Off Switch is set to "ON," but the panel does not light, you most likely have a blown mains fuse. This is included as a component of the IEC socket assembly. The lighting (LAMP or LED) is dead if the badge does not glow, but the tubes do, and there is music. It is simple to replace. The lamp resembles a fuse. A 12 volt, 0.15 amp, 1/4 X 1 1/4 light can be used to replace it. If the Stingray is upside down, it will pull straight up, but you would have to use a hook or needle-nose pliers to start it again. So while placing the Stingray, you should be careful and check the position. Back Panel Fuse Cover- To open the fuse cover, you should push in the little slot to slide the locking tab towards it. If you need to replace a fuse, you can use an MDL 3 SLO-BLOW (MDL 1.5 for 220V units). That's a sort of fuse called SLO-BLOW or TIME DELAY. Power Switch - The Power Switch is ON when it is facing towards fuse or is far from the IEC power cable, and it is OFF when it is towards the power cable. The ideal approach to start the system is to switch on the sources such as CD or turn the table first, then switch on the Stingray after 10 seconds. After a minute or so, you may turn up the volume and start playing the music. This sequence protects your speakers by preventing pops and thumps. IEC Mains Socket. The power cable passes through the IEC Mains Socket. The Rear Panel Controls of Stingray During the summer of 2005, there were many additional features added to the Stingray production version 3. These features were already available in the add-on. It must be noted that when the word input is explained, it must be taken as the stereo inputs. The stereo inputs are left and right. Everything comes in two sets. One is for the left and another for the right. Any unbalanced RCA audio analog outputs from your source components can be utilized with any of the four sets of stereo INPUT connectors. They are electrically similar, with the exception of how we designated them, which is based on recommended or regular usage. If the source you used has balanced output, you must consult the manufacturer to know the best way to connect with unbalanced loads. Using XLR to RCA converter plugs, the balanced gear can be dealt with unbalanced loads. Not only can you use these converter plugs but also specially made XLR to RCA cables. But you have to verify with every item to determine if grounding or floating Pin 3 is preferable. Transformer linked output drives unbalanced inputs by connecting XLR Pins 1&3 to the RCA shield and XLR Pin 2 to the RCA center "ho" pin. Drawback Because it's a single-ended input topology, we'd have to add chips or transformers to balance it. There is no power source for ICs inside, and there isn't enough place for extra transformers. Conclusion The Manley Laboratories Stingray is a well-made integrated amplifier with a vast, expansive soundstage, a sparkling, crystal top end, a clear, rich, and detailed middle range, and a lean, intense, articulate bass response that needs to compensate for its lack of severe low-end weight.

An introduction to the Manley VoxBox Vocal is usually the central element of a song. The human voice is foremost the natural musical instrument. Behind each note is a beautiful and complicated system. The lungs act as an air supply. A tone is produced when vocal cords vibrate. The chest and head cavities function as amplifiers. And the articulators shape the fine sound. When these work together, the result of a song is sublime. The better the vocal is recorded and mixed, the better chance for the recorded song to reach the people who hear it. The Manley Voxbox is specifically designed and optimized for the vocal recording, but its all-tube color and tone are equally based on Bass drums and other instruments used along with the vocal. Manley Laboratories is a newcomer to the audio hardware scene, having debuted in the 1990s in California. Voxbox is a channel strip. One can use the pre-built preamp, and it sounds perfect. The Voxbox plugin is a Unison enabler, and so when placed in Apollo Unison insert slot, it effectively turns the Apollo preamp into a Voxbox preamp complete with its unique gain staging tube color, custom transformer sound, and tonality of the original hardware. Voxbox has all the elements expected in a channel strip: Tube preamp, optical compressor, De-Esser Compressor, and passive EQ(Equalizer) built-in, which has a mid dip, a high peak, and low peak but it how these elements interact that makes the Voxbox so unique. The Gain control on the VoxBox injects negative feedback into the input circuits. So, it's much as tone control as gain control. Lower gain settings are neutral and transparent, whereas higher gain settings push the tone forward.  Using the Manley Voxbox, one could easily track harmonizing, rich vocals in real-time. The Manley VOXBOX was introduced in 1997 and combined the finest of Manley's audio designs into a formidable 3U compact, featuring their high-fidelity tube mic preamp, vactrol optical compressor, Pultec-style passive EQ, and de-esser/limiter. The VOXBOX is a premium tube-driven toolbox for excellent vocal tracks, including Manley's famed boutique, build quality, including in-house wound transformers.               The Manley VOXBOX Channel Strip plugin flawlessly emulates the hardware's rich, feedback-style mic preamp, chock-full of Class A vacuum tube tone. The VOXBOX plugin perfectly captures the impedance, gain staging, and circuit behaviors that have made the gear renowned, thanks to UA's breakthrough Unison technology. Unison's bi-directional control and communication are key, from the plugin to the Apollo audio interfaces' digitally controlled mic preamps.                For years, bassists have used the Manley VOXBOX as a secret weapon, with its high-end tube signal path and sleek EQ and compression appointments. Engineers have also used the Manley VOXBOX for clean electric guitars, strings, and drums because of its always-musical results, which can take any source well beyond the ordinary.  Manley Limiter The Manley Stereo Variable Mu Limiter Compressor  can be used for various applications. The unit can be used stereo or as two separate limiting or compression channels. It can be used as a balanced line amp with a gain of 24 dB! and as a low-level signal preamp. If desired, greater input gain levels can be utilized to induce gentle tube distortion. Simple settings can improve the signal in various ways that are difficult to define, but they include tube warmth, richness, greater clarity, and magic. We know some well-known producers and engineers who use LIMITER-COMPRESSORS to record all potential tracks, mix, and master them.  It's critical to comprehend the attack and recovery controls. The attack control affects the reaction to transients and percussive sounds. The time it takes for the gain to revert to normal or zero reduction is known as recovery. On certain limiters, this is referred to as "RELEASE." All-Tube Signal Path Manley has been focusing on creating superior all-tube gear since the late 1980s. In summary, Manley knows tubes, and the VOXBOX benefits from their extensive experience in tube circuit design. This channel strip was built in the United States. It has all-tube, Class A vacuum tube electronics, and the extraordinary strong build quality that Sweetwater has expected from Manley. Mic-Preamp section As its name suggests, the Manley VOXBOX is equipped with a great microphone preamplifier section. In fact, the preamp design is based on Manley's renowned Mono Microphone Preamplifier, providing you with all of the fundamentals for outstanding analog tone, whether you're using the balanced mic input, balanced line input, or 1/4" front-panel instrument input. An excellent 3-position highpass filter switch allows you to roll-off at 80Hz or 120Hz to prevent rumble, and onboard, high-current 48V phantom power may be switched from the front panel. Finally, a clever 5-position gain switch enables you to vary the amount of negative feedback introduced, allowing for a wide range of drive levels and slew rates. Change tube tonal qualities quickly and efficiently as you dial in everything from laid-back and mellow to up-front and aggressive – it's a fun way to get creative and find the right sound for each performer or instrument. Opto-Compressor The all-star compressor part of the VOXBOX was developed using features from both Manley's best-selling Variable MU and the optoisolator-based ELOP (Electro-Optical) limiter. Manley also devised a method of compressing the signal before reaching the tubes while minimizing distortion and frequency responsiveness. This implies that the VOXBOX may minimize mic-pre clipping while also eliminating entire stages of electronics, effectively shortening your microphone's standard path in half — a significant benefit. Manley also invented the sidechain, creating the "Parametric Compression" technique, which simultaneously uses up to four separate time constants and controls them using simple attack, release, and threshold settings. The Light Dependent Resistors of the VOXBOX's rapid ELOP limiter, the swift LA-2A, the slower LA-3A, and several other sophisticated and highly Manley-tuned settings are used to create the onboard presets. Overall, the VOXBOX compressor produces unique sounds that are ideal for drums, bass, vocals, and other instruments. It's even a terrific compressor for sending full mixes through if you connect two VOXBOX units together. Pultec-Style Equalizer When Manley added their Pultec-style Mid Frequency Equalizer to the VOXBOX, they took it up a notch. It now has 33 frequencies spread across three bands, including six new frequencies in the 20–150Hz range and six new frequencies in the 6.4–20kHz range. The Equalizer on the VOXBOX is equipped with actual inductors and Class-A circuitry to deliver that fantastic sound. The line input, preamp output, or insert return can all be patched in through an EQ input switch. The switch also serves as a bypass button for the preamplifier or external processors for processing two distinct tracks. De-Esser and Limiter The de-Esser and limiter sections of the VOXBOX allow you even more control over the source. Manley added a passive LC network to handle the de-essing for the peak limiter, which is based on the ELOP limiter concept. To tame troublesome frequencies, there are four carefully chosen notch frequencies: 3kHz, 6kHz, 9kHz, and 12kHz. In addition, a fifth setting provides an LA-2A-style limiter: compress before EQ, then limit after EQ. You can even insert the de-esser while the song is playing, allowing you to apply it only when necessary. Features of Voxbox Channel Strip 5 nickel laminated Manley transformers with mu-metal enclosures 2K ohm Mic input, 48V Phantom power, high-current Gain of mic preamp: 40–60dB  Gain of mic preamp: 40–60dB Input for DI instruments Line and Insert (balanced XLR, 1/4") inputs Outputs for the preamp and EQ (balanced XLR, 1/4")  XLR outputs with transformer balance 1/4" unbalanced transformerless results Compressor with De-Esser Stereo Link/Limiter  Compressor with De-Esser Stereo Link/Limiter  Sifam metre with 5 readout modes and a large lit display. 0.3 percent THD + N (1kHz @ +20dBu) +31dBu maximum output Rackmount chassis with 3U capacity. Dimensions: 19" x 5.25" x 10"

Модуль Prism Audio MDIO Dante Плата расширения MDIO Dante от Prism Sound, которая позволяет отправлять и получать звук с помощью Dante, дает вашему Atlas или Titan возможность маршрутизации каждого канала по одному проводу. Плата расширения MDIO-Dante является отраслевым стандартом для студии и аудиовизуальной сферы. Он позволяет направлять каждый чистый и яркий сигнал от AD- и DA-преобразователей Prism Sounds через CAT5e, CAT6 или оптоволоконное соединение. Он не только помогает маршрутизировать сигналы Atlas или Titan, но и помогает работать с рабочим процессом Dante, предоставляя вам доступ к сотням и сотням аудиопотоков, соединенных логическими маршрутами. Вы можете расширить возможности маршрутизации, добавив карту Prism Sound MDIO-Dante в Titan или Atlas. Особенности Prism Sound MDIO Dante 1. Расширение интерфейсов Dante для Prism Sound. 2. Специально разработан для моделей Atlas и Titan. 3. Возможна интеграция с системой Dante. 4. Поддерживает аудио Dante через CAT5e, CAT6 и даже оптоволоконное соединение. 5. Широкие возможности адаптации для интеграции с десятками тысяч каналов из различных источников. Dante — это быстрый сетевой аудиопротокол, ставший отраслевым стандартом. Он укрепил свое положение в качестве главной замены другим системам AoIP, обеспечивая сверхнизкую задержку, безупречную синхронизацию и расширяемость при большом количестве каналов. Многочисленные преимущества Dante сделали его краеугольным камнем некоторых из самых известных мировых звукозаписывающих, вещательных и образовательных организаций. Dante специализируется на обслуживании, изменении и устранении неполадок в сложных службах распространения аудио. Подключение студии или всего поста с помощью всего одного стандартного кабеля Ethernet для каждого аудиооборудования — это просто, чисто и экономично. Чрезвычайно трудоемкий и дорогостоящий процесс извлечения аналоговых или многожильных кабелей ушел в прошлое, устранив время простоя и позволив вам сосредоточиться на текущей задаче. Модуль MDIO-DANTE устанавливается быстро и легко. За считанные минуты можно вдохнуть новую жизнь и заняться маршрутизацией в студии. Простота настройки гарантирует, что единственным ограничением вашей системы будет ваше воображение. Titan и Atlas — это настоящие экспертные решения в области сетевого аудио, так что спасибо отличным сетевым функциям Dante и известному преобразованию Prism Sound. Несколько устройств Titan или Atlas можно напрямую подключить к сети DANTE с помощью модуля DANTE. Используя эту комбинацию, пользователи DANTE могут использовать аналого-цифровые и цифро-аналоговые преобразователи Titan и Atlas с превосходным качеством звука, а также их не менее высоко оцененные встроенные микрофонные предусилители. Джон Гатски из Everything Audio Network недавно назвал Titan «вершиной USB-записи» с «отличными преобразователями и микрофонным предусилителем». При подключении к DANTE управлять устройствами Prism Sound Titan или Atlas очень просто. Устройства управляются специальным программным приложением, доступным как для Mac, так и для Windows при подключении через сеть. Это позволяет управлять устройствами с любого подключенного к сети компьютера или через специальное USB-соединение. В отличие от многих сторонних интерфейсов, которые невозможно вызвать, информация о настройке может быть сохранена для быстрого поиска в специализированном Маршрутизация сигналов и конфигурация системы Dante выполняются быстро, легко и легко адаптируются. Программное обеспечение Dante Controller идентифицирует каждое устройство в сети, где маршрутизация звука, маркировка устройств и настройка сети выполняются в режиме реального времени. Эти системы чрезвычайно масштабируемы, начиная от простых соединений между консолью и компьютером и заканчивая большими сетями с сотнями и сотнями аудиоканалов. Логический маршрут позволяет расширять и изменять сети в любой момент.

The limited edition TWENTY FIVE plugin from Native Instruments Native Instruments celebrates 25 years in the industry by unveiling a KONTAKT instrument that encapsulates an array of timeless, iconic sounds from the quarter-century rich Native Instruments treasury. The library promises the addition of some exciting and innovative functionalities that are futuristic and intuitive. The plugin, part of the KONTAKT PLAY SERIES, integrates well within the KONTAKT PLAYER (6.6.1 and higher). TWENTY FIVE features sounds from thirty-seven of NI’s instruments released since its inception. From the pioneer, GENERATOR, released in the company’s induction year of 1996 to modern classics like MASSIVE X, FAR LIGHT, and MONARK, the featured instruments are undoubtedly the most popular ones loved by industry professionals worldwide. This celebratory special surprise VST could be an invaluable asset to music producers worldwide as it includes 150 cutting-edge, high-quality presets. What makes TWENTY FIVE truly special is its intuitive macro controls mapping and the presence of a powerful FX section along with a built-in sequencer. The highlight here is the fact that the instrument is designed to pair up two sound sources and then adjust, recombine/blend and re-envision an entirely new custom unique sound. In a way, this technology renders a new lease of life to classic sounds and in the process lets the future generation inherit NI’s sonic legacy while infusing contemporary sensibilities. The TWENTY FIVE interface is a stylish affair with a massive central time capsule graphic design that is easy on the eye, sleek and simple. Features that makes TWENTY FIVE truly special: There are two sound source layers called Layer A and Layer B. Upon clicking the letter A or B, the respective layer will be turned on or off. Clicking the waveform display will open the Browser displaying the entire list of sound sources available in that layer. Double-clicking on a sound source will load it and the source of the selected sound is displayed in a faded font at the bottom of the list interface. For example, the sound source of Kora is Kontakt Spotlight Collection West Africa. There are 150 snapshots (presets) segregated across 13 categories: Bass, Brass, Flute, Mallet Instruments, Organ, Piano & Keys, Plucked Strings, Sound Effects, Soundscape, Synth Lead, Synth Misc, Synth Pad, and Vocal. The six major macro controls, for example, Cutoff, Reso, Phaser, Rotator, Delay, and Reverb can all be accessed at the bottom panel from any view within the instrument interface. Balance and volume knobs are the only other parameters that also share this privilege in TWENTY FIVE. The Macro Editor is used to assign a maximum of twelve parameters to each Macro. Every existing snapshot comes with its set of unique macro assignments that can be edited as per one’s creative sensibilities. It even allows the creation of new macro assignments that come with their own perimeter ranges. The Randomize button is a thoroughly fun feature that randomizes the settings to give you an entirely new sound. The layers’ sound sources, all macros, and balance control are the parameters and settings that can be randomized. This gives a flying start to the work if one doesn’t want to sift through the entire preset list. The Sound Editor is meant for the tinkering and customization of sound sources and modulation sources respectively. The tuning, panning, volume, etc can all be controlled with the help of the sound editor. If you want to spruce things up, the randomize button can be employed to unlock new dimensions to the musical idea that the project began with. The layers can be linked, effects can be applied such as quantization noise or aliasing, filters like the brightness of the sound, etc can be rendered. The other important features of the Sound Editor include LFO controls, MOD Envelope, and AMP envelope and Modulation. FX Editor features a slew of powerful Kontakt effects for the six FX Unit slots which can be applied to both layers. It helps access the various parameters about the FX units, FX chain (the order of the effects in series for both layers) and FX presets and therefore helps control them. Layers A and B have their own Equalization or EQ for short which adjusts the levels (low, low mid, high mid, and high) of various frequencies in a signal. The user can save, load, reset and clear the FX presets for efficient management. Sequencer, accessed through the SEQ tab, recreates the functionalities of an analog sequencer. It provides seven sequencing lanes in total. The first lane is for creating a melodic sequence consisting of sixteen steps. All the next six lanes are for modulating the macro controls and their respective assignments. The various controls that apply to all sequencer lanes (Pitch/Velocity and Macros 1-6) at all times are Sequencer Preset Menu, Latch, Retrigger followed by First Step, Rate, Swing, Root Key, Scale, Global, and Sequence Lanes.Sequencing the pitch essentially is about modulating the velocity of each step during the creation of a musical phrase to unlock the dynamics of the sound. The pitch/ velocity tab has the following controls: Pitch/velocity tab, Pitch Sequencer Lane, Velocity Sequence Lane, Scale of Velocity, Layer Routing, Steps, Direction, and Gate. The controls that appear on the selection of Macros tab in Sequencer are the following: Macro 1 / 2, Macro 3 /4, Macro 5 / 6, Macro Sequence Lane, Steps, Direction, and Amount. Six knobs are dedicated to sequence macros. It is shown at the bottom of the instrument. The Settings of Twenty Five helps in controlling how it responds to the Keyboard or MIDI of the user. The elements and controls of Settings are Layer, Browser, Play mode, Velocity Curve, Note Priority, Key Range, Bend Range, Constant Rate/Time, Glide, Preserve Voices, Lock Bend Ranges. The Play Mode set of controls provides two options namely Mono and Legato. The monophonic mode sets the notes to behave like a monophonic synthesizer. The legato option provides seamless and smooth transitions between notes.The note priority settings available are Last, Low, High. The Key range allows for the highest and lowest key for the current layer. Low key and High Key settings facilitate this function.The Bend range is used to set the pitch bend wheel for both layers according to the need and preferences of the player. This function is set by the Bend Up and Bend Down settings.The Velocity Curve Shaper has various controls like Menu, Table, minimum value, maximum value, and slope. Conclusion In a nutshell, Twenty Five can be hailed as an inspiration for all audiophiles who wish to gain a deep understanding of sound design which in turn can be put to efficient use towards the creation of unique and exciting custom sounds that can provide unprecedented fresh listening experiences for audience worldwide.

An introduction to the iconic OTT plugin Music producers all around the world, especially in the world of electronic music, swear by a certain free plug-in created by Xfer Records – the OTT multiband dynamics processor. It is so popular, in fact, that one might safely presume that every electronic music producer in the world uses the OTT, or some version of it, in their day-to-day workflow. The OTT didn’t start out as a plug-in, at first. It wasn’t even originally made by Xfer Records. In fact, it was a setting in the dynamics section of the DAW Ableton Live. Originally intended as a processor for mastering, the processor could split a track into three frequency bands, and control the dynamics of each band independently, with extreme precision, using downward and upward compression (a feature which is still a rarity in the plug-in world). Multiband compression is the process of splitting an audio signal into multiple frequency regions, or bands, and compressing each band separately. When used ‘correctly’, multiband compression can maintain the broad characteristics of a signal and keep it sounding natural, while ironing out wrinkles in dynamic range across the various bands in a surgical, precise way, in order to keep everything sounding consistent. When used ‘incorrectly’, multiband compressors can be powerful tone shapers, and create unique and unexpected sounds from a source. Waves C4 and C6, and FabFilet Pro-MB are great examples of multiband compressors.      Upwards compression is a mirror image to downwards compression. Here, signals whose levels go below a certain threshold are turned up in volume, as opposed to turning down a signal when it goes above a level threshold. This is useful in mastering, because it brings out some of the subtler details of tracks when used in moderation, and can even increase the perceived loudness of a track without increasing peak (or nominal) levels. As a creative tool, upwards compression can be used to bring more clarity to the tonal part of notes, including drum, piano, or guitar sustains, or even room ambiences, and can be used to level out dynamically erratic vocal and string parts. Used excessively, however, upwards compression can increase the noise floor of a signal to a clearly audible amount, so it is recommended to use it carefully and sparingly, or on instruments that have a noise level of virtually zero, like software synths and drum machines. This is one of the reasons that the OTT is so much more popular in electronic productions than it is in productions that involve a lot of analogue equipment. The Waves MV2 is a phenomenal upwards compressor, and so are the UrsaDSP Boost, and Sonnox Oxford Inflator. OTT stands for ‘Over The Top’, and for good reason. Not only is it a multiband compressor, but each of the three bands can compress upwards as well as downwards simultaneously. As a result, when used properly, the plug-in can deliver seriously precise and musical sounds. But using too much processing, or using it without paying very close attention to minute details can easily make the effect sound over the top, and unmusical. Used too heavy-handedly, this plug-in can ruin the dynamic range of a full mix, or even individual instruments. It is a powerful tool that needs to be used carefully. Controls Let’s start by taking a look at the Time control. We’ll get to the Depth control later. The Time control adjusts the plug-in’s time constants (attack and release). Lowering the control gives you a fast attack and release, and the resultant processing squashes transients and brings up the level of tails. It can border on a distortion effect, rather than compression. Slower time constants will tend to preserve the transients, and push down the tails a little more, making the sound more percussive and punchy or thuddy than before. Usually, the most musical time constants will be somewhere in between, with snappy sounding transient information, and blooming tails that don’t sound excessively loud and overpowering, but have some movement in time with the groove of the music. On the right of the Time control are the input and output gain controls. These are fairly self explanatory. They determine the level of the signal entering and leaving the processor. The output control can be looked at as a wide band make-up gain control as well, to compensate for peak levels lost during downwards compression. The input gain control can also be crucial to operating this plug-in correctly. The threshold controls, which we shall talk about very soon, work best, and have the highest resolution when they are close to the centre of the threshold window. Setting the input gain correctly allows you to be more precise with the threshold sliders. The threshold window lies in the center of the plug-in, with six threshold controls – two for the high band, two for the mid band, and two for the low band. The brown sliders on the left hand side show us the levels at which upwards compression will be applied, and the green sliders on the right hand side show us the levels at which downwards compression will be applied. The black area in between them shows us the levels at which the audio will remain unaffected. There are blue lines and yellow dots at the centre of each of the band threshold sections. When the yellow dot appears on the right of the blue line, it indicates downwards compression being applied, and when it is near the edge, or overlapping with the blue line, it indicates upwards compression being applied. Changing the position of the threshold sliders can significantly change the tone and level of the source track, so these sliders are best used in conjunction with the knobs on the right. The knobs on the right of the threshold window are make-up gain controls for the individual bands. They, like the threshold sliders, have over 70dB of range, and must therefore be used carefully. The two controls on the bottom, labelled ‘Upwd %’ and ‘Dnwd %’ are basically ratio controls for each of the processes. This ratio applies to all three bands, so they interact very intimately with the position of the threshold sliders as well. When turned all the way down, the ratio goes close to 1:1, meaning almost no compression happens at all. On the other extreme, turning the knobs all the way up brings the ratio up a lot, getting very close to brick wall limiting. Having such a massive range of ratios makes this compressor extremely versatile, and ratios can be dialled in very precisely. However, it is for the same reason that the compressor can be easily overused. Every control on this plug-in is over the top and extreme, and can lead to imparting detrimental, or even disastrous effects on the audio signal. Therefore, all the controls must be used carefully, with care also being taken to level match processing changes as far as possible. This also brings us to the final control, on the top left of the plug-in – the Depth control. The Depth control acts as a parallel dry/wet mix. Turned all the way to the left, it turns down the level of the processed track completely, and turned all the way to the right, it exclusively puts out the processed signal. Therefore, when processing is overdone it can be reined in, or harnessed to a certain extent, by turning down the Depth control. Depending on the application, the Depth control can be very useful. Sound design processes will, more likely than not, have the depth on 100%, for instance, and mastering or group (or bus) processing will usually require some sort of parallel mix, unless processing has been applied extremely carefully. For a tool as powerful as this, the mix control is an extremely powerful feature, and makes the plug-in a lot more convenient and easy to use. The OTT is an extremely useable and powerful plug-in, that has become an industry staple, to the extent that synths like Serum, Vital, and Pigments have OTT-style compressors built into them. Slate Digital have even released their take on an OTT plug-in, claiming to improve upon its already powerful features. It is definitely worth taking this plug-in on a test drive, no matter what genre of music or sound design one works with on a regular basis. It is versatile enough to find its zone in pretty much every conceivable scenario.

An introduction to Prism Audio DSCOPE M1 Audio Analyzer The Prism Sound/Spectral Measurement dScope Series III software platform is consistent with the dScope M1 interface family. The M1 hardware is supplemented with a variety of software versions that can be bought to permit feature choices and various hardware variants. Analog Generator output values up to +26dBu are supported. The Analog inputs with a sampling rate of up to 768kHz and a bandwidth of more than 300kHz are supported. There are two-channel digital Input and output(AES3 and S/PDIf modes) The streaming design offers for more flexibility in creation and analysis, as well as faster processing. The most basic electro mechanical components are used for lengthy service life. 12V AC-DC power converter Small size: 36x230x180mm, rack-mount ears available. All standard audio tests and more with easy computerization and reportage in a compact and lightweight package You can use the software with ease Simple "Auto Sequence" "code-free" automation based on your previous configurations or write your own automation in VBs, Python, C++/C#, or another language. The Bandwidth from DC to >200kHz; 100V RMS maximum input; THD+N residual typical -110dB There is an automation option with +48 phantom power, and it also provides the excellent feature like any other professional equipment The Tetrahedral Test Chambers and bespoke products that can completely surround a speaker cabinet or Head and Torso Simulator (HATS) and create balloon responses and 2*Pi measurements are available from Hill Acoustics Loudspeaker Test Chambers. These eliminate the need for costly anechoic chambers and give a precise positioning jig that ensures very reproducible results. Features of the dScope M1 The Analog Input and output (48kHz/96kHz/192kHz) The Digital Input or output is set to a maximum of 192kHz The Sound Card I/O uses ASIO/WDM devices The other supported devices are Bluetooth, Dante, and MADI The Amplitude, Frequency, and Phaseanalyses the signal Sine, White/Pink noise, Swept sine, Twin-tone generated the signal Basic AES3 is the status of the channel. Measurements of time and frequency domain. THD+N, THD, individual/odd/even harmonics, IMD (CCIF / SMPTE-DIN) Cross-talk, Balance, Gain, Noise Standard and user weighting filters There is FFT Analysis with Smart FFT Detectors. The Sweeps are another unique feature. An impulse Response Test can be done. Scripting Code freeThe test automation of code is free, and the reports can be generated using Excel, PDF, or HTML. The External control is done through ActiveX like C++, Python, LabView, Matlab, etc. From designing the product through production line QC testing, Prism Sound's audio test solutions are appropriate for automotive test applications. All components of the In-Car Entertainment (ICE) system may be examined with a single instrument. You may use the audio devices to send test signals straight from dScope III to your PC's Bluetooth chipset or the EUT's USB dongle. It would be best to connect directly between the test system and Bluetooth device for your Bluetooth system tests. This will give instant way into the distinctive Scope III feature set, which includes sweeps, regulation, Smart FFT detectors, live measurement traces, multi-tone analysis, automation for QC applications. The most significant benefit of adopting dScope Series III for Bluetooth testing is that it eliminates the requirement for tedious wave file generation and processing in external software. Finding audio faults in a broadcast chain is more difficult today than it has ever been. For analog signal chains, the tried and true approaches no longer apply. With this in mind, the dScope and DSA-1 were created in collaboration with significant broadcasters and their standards bodies. They have measuring capabilities that are not available on any other equipment. Built-in automatic analysis tools, such as a watchdog event recorder for detecting and catching those pesky clicks and pops, as well as ways to look at and analyze the digital stream, give capabilities that no other instrument can match. Prism Sound's dScope Series III audio test system is known as the world's first electronic and acoustic test system which can address the Windows® audio devices. The dScope Series III combines both software and hardware analysis systems that analyze Windows Sound devices as if they were any other input or output (WDM or ASIO). It may either direct to a sound card or is measured by using a sound card as input, and it would do cross-domain, analog or digital, input or output. Prism Sound provides comprehensive system models and examination capabilities for electro-acoustic transducer design. The LOUDSOFT software modeling suite allows engineers to build and simulate a full loudspeaker from the motor and soft components of the transducer using the inclusion and cross-over design. The dScope Series III Audio Analyzer is the only instrument in the world that can assess a complete electro-acoustic system, including loudspeakers, microphones, integrated analog and digital amplifiers, A/D, and D/A converters, and a Windows® audio communication link that includes USB and Bluetooth. Prism Sound delivers audio measuring solutions for the whole range of consumer audio and home entertainment goods, which is ever-diversifying. The product design right through to the production line, the dScope Series III test and measurement platform deliver inclusive test routines from media players to Bluetooth speakers, from portable FM and digital radios to reference-quality home cinema schemes, from noise-stopping headphones to mobile telephones. For production line / QC testing applications, the dScope Series III is one of the fastest and most capable audio test options. With the ability to concurrently measure over 100 audio parameters and sophisticated FFT-based analysis features like multi-tone testing and impulse response analysis, the dScope Series III may help you enhance throughput and verify that you're shipping fault-free goods. The dScope Series III Audio Analyzer has unrivaled capabilities for evaluating mobile audio devices such as MP3/Media players, earbuds, and headphones. Measurements can be taken in accordance with industry standards, such as the British Standard BS EN 50332 for the hearing safety of portable music players. Furthermore, up to 40+ performance metrics, such as warp, sound, frequency response, gain, signal-to-noise, ratios of odd to even harmonics, and dynamic range, can be measured and analyzed in real-time using the analyzer's real-time interactive measurements and unique Smart FFT™ capability. The dScope analyzer can readily adjust for the ear's frequency response characteristics when using ear simulators, assuring reliable measurement results of earbuds and headphones. Although the dScope Series III audio analyzer is a soft instrument, it may be utilized to obtain those difficult-to-obtain findings. Automation may be used to simplify complicated processes and make the impossible feasible in several ways. The user-defined FFT detector may be designed to yield a result that can be utilized just like any other FFT detector result: swept, limit tested, used in regulatory algorithms, and so on. So, Prism Sound's audio test and measurement solutions were created to help the company design and construct the highest-quality, reference-grade studio recording interfaces. However, they are now utilized worldwide to assist other manufacturers and designers in developing their own products. Engineers can test their goods more thoroughly and quickly than ever before with no other audio test and measurement solution. Engineers can test their products more extensively and promptly using no additional audio testing and measuring solution. Prism Sound is the industry leader in semiconductor audio measurement and test equipment. You must be able to access the audio at the board level to accomplish this. The VSIO adaptor for the dScope Series III audio analyzer makes it simple to achieve this with digital audio. It's the only multiformat I2C capable multichannel digital serial interface on the market, with 16 channels of audio data on a single wire and a dedicated control connection for I2C or SDI serial transmission. This is excellent for every chip-level audio design need. The VSIO can handle up to 16 multiplexed channels on a single wire, 8 on each of two wires, or four on each of four lines without any additional hardware.

Учебное пособие по реверберации для лупов и семплов | Учебник по музыке Реверберация — это видимое постоянство звука, слышимое, когда звук отражается от твердых поверхностей к ушам слушателя. Нас постоянно окружает реверберация или эхо, и наши уши очень чувствительны к ним. Хотя в повседневной жизни мы редко замечаем реверберацию в комнате, ее отсутствие может быть поразительным, а иногда даже раздражающим. За прошедшие годы реверберация стала неотъемлемой частью музыкального производства и используется по-разному: от придания источнику звука ощущения пространства до уникальных творческих эффектов и сюрреалистической атмосферы. В этой статье мы рассмотрим несколько плагинов, которые эмулируют различные единицы реверберации, пространства реверберации и креативные эффекты реверберации, которые сегодня нашли прочную основу в мире музыкального производства.