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• FTA (Final Touch Audio) Callisto USB cable
• GOTEBORGSAVENTYR
• Microphones, Speakers & Audio Processing
• US20120033829A1 - Audio phase corrector - Google Patents
• Audio Phase Converter ASC 330vac 40uF Capacitors LOT OF 4 polypropylene Film
• Best active speakers 2021: floorstander, desktop, budget and premium
• Understanding Audio Phase
• XAudio2 Volume and Pitch Control
• shop sale discount Ultrasport Audiophase DM9915 CD PLAYER great deals
• The Dark Night Rises

WATCH RELATED VIDEO: Feniks ESSENCE: the world's best computer speaker system (Kickstarter video)

FTA (Final Touch Audio) Callisto USB cable

An audio phase corrector circuit which shifts the phase of substantially all frequencies of an audio signal by 90 degrees for the purpose of enhancing sound produced by an electroacoustic transducer. Embodiments are implemented by a digital finite impulse response filter configured to operate as a degree phase shift circuit, commonly called a Hilbert transformer. The degree phase shift corrects a characteristic phase distortion caused by electroacoustic transducers and thereby retains much of the impulse and transient information in the acoustic output that would otherwise be lost.

The resulting sound has substantially improved clarity, detail, presence, placement, and spaciousness. The present invention generally relates to the field of electronic audio equipment.

More particularly, the present invention relates to the field of audio signal conditioning circuits that enhance the quality of sound produced by electroacoustic transducers such as speakers, headphones, and hearing aids. Audio signal conditioning circuits that enhance the quality of sound are hereafter referred to as audio enhancement circuits. Such circuits operate by altering an electrical signal that is an analog or digital representation of acoustic sound waves.

The purpose of altering the signal is to cause an electroacoustic transducer to produce a sound that is perceived to be superior to the sound it would otherwise produce when driven by the unaltered signal. In general, a prior art audio enhancement circuit alters an audio signal by selectively changing its amplitude or phase characteristics or by adding new frequencies to the signal.

Usually these changes and additions are attempts to compensate for poor acoustic transient response by bolstering the amplitude and rise time of signal transients driving the transducer. Such attempts are justifiable because good acoustic transient response is necessary for producing the natural shape and frequency content of sound impulses, especially for percussive and vocal sounds. It appears that some implementations of the prior art have achieved modest to significant success if a comparison to their degree of commercial success is valid.

But the prior art has significant shortcomings that have kept it from becoming more broadly applicable to the field of electroacoustic sound production. This is because, unlike the present invention, no implementation of the prior art consistently enhances sound without requiring adjustments for different operating conditions or without altering the program content of the original signal.

Specific examples of prior art as related to the present invention are presented in the following paragraphs. The various embodiments of U. Adjustments are required for different operating conditions. The output signal is amplitude shaped as a function of the input signal's time and amplitude envelope. A class of widely used audio enhancement circuits consists of tube amplifiers, which add harmonics to the signal, especially when they are being mildly overdriven toward saturation.

The resultant gentle rounding of transient signal peaks produces even harmonics that enhance the resulting sound in the minds of some listeners, including some that prefer and can afford high-end audiophile equipment. The appeal of tube amplifiers is based on a psychoacoustic effect produced by the added harmonics. But by adding harmonics, the tube amplifiers do indeed produce a form of distortion to the original signal. There are a variety of audio related circuits in the prior art that do not enhance sound, but, for different purposes, they use a signal processing function that is also used in embodiments of the present invention.

Said signal processing function implements an electronic filter that phase shifts substantially all frequencies in an audio signal by 90 degrees. Such a filter is commonly referred to as a Hilbert transformer. Two common uses of the Hilbert transformer that are representative of many similar uses in the prior art include encoding and decoding duplexed audio signals in surround sound systems and voice modulation and demodulation of single sideband carriers in radio communication systems.

Other prior art uses involve esoteric analysis and manipulation of complex audio signals. A representative example of esoteric analysis and manipulation is U.

It is important to note that there are other circuits and software functions called Hilbert transformers that are used in the prior art, but they are not used to enhance sound in the manner of the present invention.

Common examples of these are digital infinite impulse response IIR Hilbert transformers and analog allpass Hilbert transformers. Hilbert transformers such as these produce two signals in quadrature in which each frequency in one signal has a degree phase difference with the same frequency in the other signal, but the relative phase of all the frequencies in the two signals are scrambled with respect to the original signal.

The resulting phase scrambling is counterproductive to sound enhancement. A theory of the art, as it relates to the prior art and the present invention, is presented in the following paragraphs. The theory describes a characteristic phase distortion produced by typical electroacoustic transducers and explains why the prior art fails to consider and correct the characteristic phase distortion.

Also, it explains how the present invention corrects the characteristic phase distortion and presents a rationale for why the present invention thereby enhances sound.

Since the efficacy of the present invention has been otherwise verified by listening tests, be advised that the validity of this theory is not a prerequisite to establishing the merits of the present invention. The prior art is mostly based on the premise that performance of an electroacoustic transducer, such as a speaker, can be enhanced by altering an audio signal to compensate for the transducer's distortion inducing inductance, capacitance, reverse electromotive force, inertia, friction, reflections, resonances, cone or diaphragm breakup, and vibration modes.

In simple terms, if a hypothetical electrodynamic or electrostatic speaker produced no phase distortion other than that of its characteristic transfer function, in accordance with FIG. The dynamics of this source of phase distortion also applies to non-continuous signals such as impulses and transients. Any other phase shift is primarily caused by speaker cone or diaphragm breakup, vibration modes, and resonances. Usually speaker cones are intentionally designed to have breakup and vibration modes in order to broaden their frequency response, but the unavoidable consequence is additional phase distortion.

Other distortion factors of real speakers contribute primarily to harmonic and amplitude distortion, but they can also contribute to phase distortion. The need for the correction is based on a fact of physics that acceleration of a mass is proportional to the force exerted on it.

This means that if a voice coil driven cone or a diaphragm motion is to correspond to an audio signal's original shape, zero signal voltages need to be transformed to peak voltages in order to produce the forces that cause the high-velocity zero crossing motion required of continuous signals and the fast leading edge motion of impulses and transients.

Peak signal voltages need to be transformed to zero voltages to produce zero forces that stop motion. Signal voltages between the zeroes and peaks need to be transformed to values that produce proportional forces of the proper polarity for the motion required between the zeroes and peaks. The resulting acoustic pressure waves should correspond to the original uncorrected signal voltage and the displacement of the cone or diaphragm.

Verifying this theory with full bandwidth acoustic measurements is not easy to accomplish because of frequency group delay and the phase scrambling nature of real speakers and measurement microphones. However, the theory is bolstered by the results of a relatively simple phase shift measurement test of a speaker driven by an audio input signal consisting of two phase coherent frequencies.

The phase shift measurement test setup is illustrated in FIG. In the test setup, an adjustable sine wave oscillator is set up to produce a continuous single-frequency sine wave that drives a dual-frequency generator consisting of a frequency doubler and a summing amplifier The outputs of the frequency doubler and the sine wave oscillator are summed by the summing amplifier to produce a basic test signal The basic test signal is composed of two equal amplitude frequencies that are one octave apart and have a fixed phase relationship.

A measurement microphone converts the acoustic output of the speaker to an electrical signal, which is boosted by a preamplifier and displayed on an oscilloscope An uncorrected speaker output and a Hilbert-corrected speaker output are displayed on the oscilloscope and should appear as shown in FIG. Note that the Hilbert-corrected speaker output appears the same as the basic test signal Using this test setup to measure phase response of a typical speaker's characteristic transfer function will usually produce a combination of supporting, conflicting, and ambiguous results at various output frequencies of the sine wave oscillator.

This is because of the phase scrambling nature of most speakers. In order to mitigate such results, a test speaker was chosen for its low inductance and stiff cone to minimize phase shift other than that of the characteristic transfer function.

Test frequencies were chosen to avoid resonances and vibration modes that would significantly distort amplitude and phase response. Even so, the test speaker had a relatively narrow band that was not significantly affected by other sources of distortion. An unavoidable question is, why does phase correction of an electroacoustic transducer's characteristic transfer function enhance sound in spite of all the other sources of phase distortion?

The primary reason is that the characteristic phase distortion is the most pervasive and consistent form of phase distortion. By correcting frequency phase alignment for this form of distortion, much of the original signal's impulse and transient response is restored in the acoustic output of the transducer. This helps to restore the resolution and directivity information of the original signal. The general result is more clarity and detail in the sound.

For stereophonic and multichannel systems there is a greater sense of presence, placement, directivity, and spaciousness. The ability of the present invention to enhance audio is limited only by any overriding influence of distortion from sources other than a transducer's characteristic transfer function.

The present invention enhances sound in a way that is simple in principle but technically sophisticated to implement and not intuitively obvious, which is probably why it was not previously anticipated. Unlike the prior art, it accomplishes its purpose without adding new frequencies to the original signal, without altering amplitude and phase characteristics of selected portions of the original signal, and without requiring adjustments for different operating conditions.

Instead, the present invention shifts the phase of substantially all frequencies of an audio signal by 90 degrees for the sole and novel purpose of enhancing the sound produced by an electroacoustic transducer. An embodiment of the present invention has been built and operated in combination with conventional audio signal sources, amplifiers, and electroacoustic transducers to produce sound that is significantly enhanced, especially for stereophonic and multichannel sound systems.

Although phase shifting all frequencies by 90 degrees may seem an unlikely way to enhance sound, its efficacy has been thoroughly verified by the positive opinions of listeners.

The sound improvement is initially subtle because there is virtually no change in program content. But as the brain quickly adapts to the degree phase shift phenomenon, the listener notices a robust improvement in clarity, detail, presence, placement, directivity, and spaciousness. It is as if a veil has been removed and the sound has been transformed from two dimensions to three dimensions. Poorly produced program content is not transformed into good program content, but most program content is improved relative to its listenability.

With respect to the prior art, it is possible that various implementations of it could be used in combination with the present invention to further compensate for perceived deficiencies in audio signals and equipment.

A broadly applicable and readily implemented form of the present invention is a finite impulse response FIR filter that operates in the digital domain. The FIR filter is configured to perform the function of the degree phase shift circuit, which is commonly called a Hilbert transformer. The Hilbert transformer is a standard signal processing function that can be performed by a variety of hardware, software, and firmware implementations. With conventional input and output interface circuits the present invention can be integrated into any analog or digital context of an audio signal chain, from a program source to a power amplifier, either as an autonomous unit or imbedded within another unit of audio equipment.

One embodiment is a DSP-based Hilbert transformer used specifically to enhance sound produced by electroacoustic transducers while operating in an analog-to-analog context of an audio signal chain. With reference to FIG. The embodiment described herein has two signal channels, but it can be implemented with any number of signal channels. It can also be implemented with other functionally similar custom made or commercially available hardware. The PGAs a and b buffer and scale analog audio input signals a and b.

The LDAs a and b buffer and scale the outputs of the DACs a and b to produce analog audio output signals a and b. A software program for the DSP engine consists of a standard time-domain convolution FIR filter main routine and supporting software routines and drivers for the input and output interface. A coefficient data table is used by the FIR filter main routine to invoke the Hilbert transformer function.

In this form of the embodiment a data table of Hilbert coefficients with three added zeros for each channel is stored in DSP memory space for a Hilbert coefficient table and used by the tap FIR filter main routine to achieve maximum frequency bandwidth with the DSP engine operating at a 96, Hz sample rate and a MHz instruction rate. Frequency response is flat from 20, Hz down to 40 Hz and is down by 2. The voltage gains of the PGAs a and b , the LDAs a and b , and the coefficients in the Hilbert coefficient table should each be scaled to produce the desired net gain while preventing signal clipping.

Of the many resources for calculating Hilbert coefficient data tables, two commonly used ones are Iowegian's Scope FIR interactive filter design software, which can be downloaded from the interne, and PTC's Mathcad signal processing software. Also, Hilbert coefficients can be calculated in a manner describe by Lyons See R.

PC application software such as Scope FIR and Mathcad can generate or import Hilbert coefficients and graphically display their frequency and phase response for evaluation and amplitude scaling.

To accomplish this in this particular form of the embodiment, the Hilbert coefficient data should be arranged with negative coefficients preceding positive coefficients. An alternative implementation of the Hilbert transformer function in software is a frequency-domain fast fourier transform FFT FIR filter main routine that can broaden the frequency bandwidth by adding approximately an octave or more of low frequency response for a given set of DSP hardware and software assets such as those described above.

GOTEBORGSAVENTYR

Albedo and Audio Connect had a couple of rooms at the show. Speakers in both rooms. Buchardt loudspeakers being driven by an unknown B amplifier and Cocktail Audio streamer. Cables in the room were from Tellurium Q. The speaker model is the S which is a two way system with a passive radiator round the back.

Microphones, Speakers & Audio Processing

As for this particular project, the Sound Sense team knew The Dark Night project in Anand was one that would test its creative skills when a local business owner called about his needs. The high-end electronics and home appliances showroom owner wanted a unique, discrete, world-class home cinema set-up with a large screen and a seating capacity for seven people. The cinema set-up would require audio and video calibration and one-touch control. The cinema space, intended to provide a welcome break from work time, is conveniently located adjacent to the store showroom and office. The existing room structure needed updates to make it a viable home cinema space. Audio isolation was a top priority while still providing an outstanding cinema experience. The entry to the cinema set-up was from the rear, center of the room.

US20120033829A1 - Audio phase corrector - Google Patents

The most common design is a so called 2-way system. That means, that the frequencies are produced by two transducers, each specifically designed for a certain frequency range:. Generally, active studio monitors are equipped with an internal amplifier and signal processing filtering, EQ-ing whereas passive monitors rely on an external, additional device. Active studio monitors allow a low-loss transmission of electric signals from a sound source PC, interface to their input connectors maintaining linear characteristics. This is one of the most frequently asked questions and a common misunderstanding.

Audio Phase Converter ASC 330vac 40uF Capacitors LOT OF 4 polypropylene Film

JavaScript seems to be disabled in your browser. You must have JavaScript enabled in your browser to utilize the functionality of this website. You might be experiencing phase cancellation, a phenomenon that can make certain frequencies vanish from your mix. To help you out, this Studio Basics article will help you understand phase — what it is, why it matters, and what it means to be out of phase. Essentially, phase refers to sound waves — or simply put, the vibration of air.

Best active speakers 2021: floorstander, desktop, budget and premium

Quick physics lesson: speakers can be categorised into two different species — active and passive. Passive speakers are your 'standard' driver-filled boxes that require an external amplifier to make a sound. Once the signal from the amplifier reaches these boxes, there's an internal, non-powered crossover built-in to filter the appropriate frequencies to each of the drivers. Stay with us: there are also speakers that are passive in nature, but still have an amplifier built-in — often squirrelled away into one of the boxes. These beasts are referred to as 'powered', and are represented by most of the sub-thousand-pound products on this list. Now, active speakers.

Understanding Audio Phase

It wasn't fair because his was a 7. Augmenting that grouping were four JL Audio "Gotham" active subwoofers placed around the room, plus a Sony 4K projector and a " Screen Innovations Black Diamond screen the photo doesn't begin to do this set-up justice. Oh, and there was also a UnitedHome Audio Phase 11 reel to reel tape deck and some very high quality tapes. You needed a ticket for the demo and it was well worth the wait.

XAudio2 Volume and Pitch Control

RELATED VIDEO: Audio Phase: Why You Need To Pay Attention To This

Many people get confused when trying to determine what phase to set their subwoofer to as well as how to tell if their subwoofer is in or out of phase. In this article, we will simplify the idea of audio phase as well as explain how to tell if your subwoofer is out of phase. Simply put, audio phase refers to the timing of the audio waveform or audio signal. Audio signals travel as what we call sine waves which comprise of peaks and troughs. If two waves are timed exactly the same, their peaks and troughs will match up and the waves are said to be in phase. Since the waves are in phase and the peaks and troughs match up, the signals with add up.

shop sale discount Ultrasport Audiophase DM9915 CD PLAYER great deals

Reversed phase in audio can happen when audio devices save audio in the wrong way, resulting in a bad audio, notwithstanding everything seems to sound fine with the headphones. A reversed phase audio creates problems on certain devices. A tell tale could be that the sounds appears to be thin, no bass, etc. Luckily, there is a good way to find out, using Audacity , or any other audio editor. When you open your audio in the editor, a good audio will show you something like this:. As you can see, the wave pattern of this talking audio looks more or less the same, with some small variations.

The Dark Night Rises

The OR uses a x high contrast ratio panel that can display high-quality images from any source or aspect ratio. This unit can be used as a portable stand alone monitor equipped with front panel stereo speakers, integrated carrying handle, and desktop stand. An optional rack mount kit with tilt capability is also available.