TheAshkOne - All Blue (Original Mix)

How ribbon microphones work?!

1. Ribbon consist of an extremely thin aluminium foil placed between two neodymium magnets which vibrates with sound waves, The movement of the ribbon in the magnetic field generates a small electromotive force (emf) which can be detected using two wires connected to the ribbon.

2.This signal is so weak so we use a transformer to step-up the voltage.

3.The output of the transformer is now ready to send through XLR cables to preamps.

History and technical development of the ribbon microphones [1932-2016]

The ribbon microphone was patented in 1932 by Harry F. Olson. It consists of an extremely thin aluminium foil ribbon suspended in a magnetic field. Pressure gradients in the air cause the ribbon to move. The movement of the ribbon in the magnetic field generates a small electromotive force (emf) which can be amplified and recorded. The ribbon microphone quickly gained popularity with audio engineers for its uniform frequency response.

When magnetic tape became the dominant recording media, ribbon microphones became less popular and condenser microphones took over. With recordings and sound mixing processes making use of magnetic media there is always a slight loss of high frequencies. This problem could be remedied by large capsule condenser microphones. These microphones have a number of resonances in the 8 kHz to 12 kHz range that enhances the high frequencies before recording. The capsules of condenser microphones are tensioned tightly, causing the high-frequency resonances. 

The aluminium foil element of the ribbon microphone is only lightly tensioned, causing resonance at very low frequencies. When digital recording became the order of the day, ribbon microphones made a major comeback because high-frequency transfer loss was no longer an issue. Its ability to record fast transients accurately without adding upper-range resonances became again a very positive attribute.

Articles about the ribbon microphone dates back as far as 1931. Most of the early publications were authored by Harry F. Olson who patented the ribbon microphone. Few publications besides those by Olson can be found about the ribbon microphone, but plenty of patents relating to the ribbon microphone are freely available. A selection of those patents is discussed in the paragraphs to follow.

Olson 1932 

Harry Olson filed the first patent for the ribbon microphone in 1931 and the patent was awarded on 25 October 1932 with the title “Apparatus for converting sound vibrations into electrical variations”.

The patent illustrations are of very sturdy mechanical designs as illustrated in picture below:

It describes the ribbon as a relatively small item that is supported in such a way that it resembles the motion of a particle in free air. A device of this nature is classified as a velocity microphone. The combination of mechanical parts surrounding the ribbon is called a baffle. The size of the baffle around the ribbon is calculated according to the highest frequency that the microphone is designed for. The baffle should be designed so that the path length from the front of the ribbon to the rear of the ribbon is half the wavelength of the required frequency. Picture below provides a graph from the patent to illustrate the effect of the baffle size on the frequency response of the microphone. 

The ribbon is made of conducting material that is light in weight and that has low elasticity, for example aluminium foil. The ribbon must not be stretched tightly between its supports. It is crimped in order to suspend it rather loosely between its supports to promote flexibility along its whole length. The supports are made of non-ferro-magnetic conducting material, but it is electrically isolated from the ribbon by non-conductive material. The two signal wire leads are electrically connected to the two ends of the ribbon. The light weight and small restoring force of the ribbon causes its natural vibration frequency to be below the audible range. Tests that were done before the patent submission had shown that a natural vibration frequency of approximately 10Hz produced the most desirable results. The patent states that “When a diaphragm of small mass is suspended in this manner its mechanical reactance is small compared to the impedance of the air. In other words its mass reactance is negligible over a large frequency range compared to the acoustic resistance of the air which it displaces.” 

The ribbon is suspended in the air gap between two poles of a magnet in such an orientation that its surfaces are parallel with the magnetic force lines. The magnet can be a permanent magnet or an electromagnet. The gap between the ribbon and the magnet poles are kept to a minimum to prevent the leakage of air around the ribbon, but the gaps must still be sufficient to prevent frictional contact between the ribbon and the magnet. The patent suggests an air gap of 5.6mm with the ribbon slightly narrower. The ribbon cuts the magnetic field lines while moving in the air gap between the poles because of the sound pressure variations across it. This causes an electromotive force that is proportional to its movement. The electromotive force can be amplified with suitable electronic equipment.  

The magnetic pole pieces with its supporting structure forms the baffle. The baffle increases the path length from the front to the back of the ribbon. The length of this path has an influence on the response of the microphone. The paths around the top and bottom supports of the ribbon are shorter than the paths around the baffle, but its effect is relatively small because it influences only a small part of the ribbon. The clamping structure that secures each end of the ribbon is made of non-magnetic material (ex. copper or brass if it is made from metal). Although it is desirable to make the ribbon as light as possible, it is sometimes necessary to vary its thickness in order to increase its efficiency within the particular baffle setup. The movement of the ribbon is caused by the phase difference of the sound wave between the front and back of the ribbon. The phase difference is determined by the distance that the sound wave has to travel around the baffle from the front to the back of the ribbon. The greatest phase difference occurs when the path length around the baffle is half a wavelength of the sound wave under question. Olson provides a helpful visual representation of this effect in his patent. Picture below shows a horizontal cut through the microphone and a sinusoidal representation of a sound wave. If the path length A-B around the baffle is plotted as E-F on the axis of the sound wave, then the pressure difference between points A and B on the ribbon will be equal to the sum of C-E plus D-F.

The sound intensity at the opposite sides of the baffle is virtually the same for all wavelengths that are longer than twice the distance around the baffle. At wavelengths shorter than this, the intensity on the approaching side of the ribbon increases and the intensity at the retreating side decreases. The reasoning follows that the pressure difference across the ribbon is proportional to the frequency as long as the distance around the baffle is less than half the wave length. Due to the nature of the design the ribbon microphone exhibits very directional characteristics. Sound waves coming from an angle will produce less of a pressure difference across the ribbon. A sound wave directly from the side will produce virtually zero pressure difference. The pressure difference can be calculated with simple trigonometry rules. An alternative design is also provided in the patent. The alternative design does not make use of a ribbon, but of a lightweight diaphragm. This design is however overly complicated and will not be discussed further. 

Olson & Weinberger 1933 

In 1933 Olson, in cooperation with Julius Weinberger, filed a patent that made use of a combination of the pressure gradient (velocity) ribbon microphone and a pressure component microphone to achieve unidirectional operation. Unidirectional operation is desirable in order to improve the ratio of the sound source relative to the sound reflections in the room. The patent achieved its purpose through the use of a normal ribbon (as in the 1931 patent) assembled in series with a modified ribbon. The modified ribbon was adapted to act as a pressure microphone by enclosing the back of the ribbon. The normal and modified ribbons are working in phase for sounds generated in front of the microphone, but are out of phase for sounds originating from the rear side of the microphone. This patent illustrates the influence of the baffle in the extreme case of making the baffle infinitely large by enclosing the rear side of the ribbon completely. 

Anderson 1937 

Leslie Anderson added electronics to Olson and Weinberger‟s unidirectional microphone to file a patent in 1937. The electronics made it possible to adjust the phase differences between the two ribbons, thereby making it possible to adjust the directionality of the microphone from the mixing desk.

Ruttenberg 1938

In 1938 Samuel Ruttenberg filed a patent to address one of the imperfections of the ribbon microphone. When speaking close to the microphone, the low frequencies are over emphasised because the higher frequencies are attenuated. The high frequency attenuation happens because different sections of the ribbon move out of phase. Thus one section cancels out the electrical current that is generated during vibration of another section. The sections move out of phase due to the fact that the ribbon is longer than the wavelength of the higher frequency sound waves. Ruttenberg addressed this problem by designing a special housing for the microphone which closes up the rear of the microphone with an adjustable shutter. When closing up the rear of the microphone, its characteristics are changed from that of a velocity microphone to that of a pressure microphone because the sound wave does not have access to the rear of the ribbon. This idea clearly borrows from the same principles than Olson and Weinberger‟s patent on the unidirectional microphone. The microphone‟s operation in pressure mode tends to minimise the effect of high frequencies being attenuated. This patent clearly illustrates that tampering with the physical surroundings of the ribbon microphone, does have a distinct influence on its operation.

Bostwick 1938

Telephone conferencing has been around for longer than one would expect. In 1938 Lee Bostwick already addressed the problem of feedback between speaker and microphone during teleconferencing. He filed a patent for a device making use of two ribbon microphones and a loudspeaker. The device is constructed with a loudspeaker facing downwards onto a deflector underneath that reflects the sound waves in a horizontal direction toward the conference attendees. Two ribbon microphones are fitted perpendicular on top of each other, both on top of the loudspeaker. Because of its directionality, the ribbon microphones are insensitive to the sound waves emanating from below it, but are sensitive to the voices of the conference attendees around the table.

Bostwick‟s teleconference solution

Olson 1940

In 1940, Harry Olson filed a patent for an improved version of the 1933 unidirectional microphone . Olson discovered that he can design a unidirectional microphone with a single ribbon instead of the dual-ribbon method used previously. Furthermore, this microphone could be easily changed from unidirectional to bidirectional, to non-directional operation. The method of achieving this was to place a pipe structure (resemblance of a smoking pipe) behind a single ribbon. The pipe ends in a labyrinth that is filled with a soft fabric acting as acoustic resistance. The pipe has shutters that can be opened or closed to achieve the desired results, i.e. shutters open allows bidirectional operation, shutters partly closed enables unidirectional operation and shutters completely closed constrains it to non-directional operation. This patent illustrates once again the influence of the surroundings of the ribbon on its operation.  

Anderson 1942 

Leslie Anderson built upon Olson‟s 1933 patent by filing a patent in 1942 about the magnetic equalization of sensitivity in a unidirectional microphone. It is fundamentally an improvement on Olson and Weinberger‟s design by adding a mechanism to adjust the magnetic fields for the two ribbons and by changing the shape of the pole pieces in such a way that it is possible to vary the flux density of one air gap relative to the other. This patent indicates support for Olson and Weinberger‟s idea of modifying the baffle in order to change the operation of the microphone for a specific purpose.

Rogers 1942 

Ernest Rogers filed a patent in 1942 for a microphone with selective discrimination between sound sources . He claimed that the microphone can be used to determine the direction of a sound source. The microphone receives sound waves approaching the microphone straight-on, but attenuates sound waves approaching the microphone from an angular displaced direction. The construction of the microphone is basically four ribbon microphones assembled in an X-pattern. The ribbons are connected to a mixing circuit in such a way that the phase relationship between the ribbons can be adjusted. By adjusting the phase relationship, the microphone can be tuned so that the phases of sound waves from a certain direction cancel each other out, while the phases of sounds waves from another direction will add to each other. 

Olson 1946

The ribbon microphone inherently has a directivity pattern. Olson filed a patent in 1946 to combine two ribbon microphones in a perpendicularly fashion in order to get a microphone that has a 360° pattern on the horizontal plane. This microphone would be ideal for use in orchestras for example, recording all the instruments around it, but attenuating sound waves reflecting from the ceiling and from the floor. Part of the idea is borrowed from Bostwick‟s teleconferencing microphone discussed earlier. 

Anderson 1947 

Anderson‟s patent application in 1947 was filed as an improvement on Olson‟s 1931 design. According to Anderson, Olson‟s design displayed a considerable drop in output when the distance around baffle approaches one fourth of the sound wave‟s wavelength. Anderson‟s design claimed to have a more uniform response over the operating range of the microphone and also to have an enhanced high frequency response compared to the conventional design. This was accomplished by mounting one or more semi-circular bands behind the ribbon. The bands provide cavities that are resonant at the frequencies where the enhancements are deemed necessary.  

Nature of phaselock

A phaselock loop contains three basic components

1. A phase detector (PD).
2. A loop filter
3. A voltage-controlled oscillator (VCO) , whose frequency is controlled by an external voltage.

The phase detector compares the phase of a periodic input signal against the phase of the VCO; output of the PD is a measure of the phase diffrence between its two inputs. The diffrence voltage is then filtered by the loop filter and applied to the VCO, Control voltage on the VCO changes the frequency in a direction that reduces the phase diffrence between the input signal and the local oscillator.
When the loop is locked,the control voltage is such that the frequency of the VCO is exactly equal to the average frequency of the input signal.For each cycle of input there is one, and only one , cycle of oscillator output.
One obvious application of phaselock is in automatic frequency control (AFC). Perfect frequency control can be achieved by this method,whereas conventional AFC techniques necessarily entail some frequency error.
To maintain the control voltage needed for lock it is generally necessary to have a nonzero output from the phase detector.Consequently , the loop operates with some phase error present ; as a practical matter, however this error tends to be small in a well designed loop.

TIP #1 - Effect of sound level on the perception of pitch

The level of sound affects the perception of pitch. For low frequencies, the pitch goes down as the level of sound is increased. At high frequencies, the reverse takes place—the pitch increases with sound level. The following is an experiment within the reach of many readers that was suggested by Harvey Fletcher. Two audio oscillators are required, as well as a frequency counter. One oscillator is fed to the input of one channel of a high-fidelity system, the other oscillator to the other channel. After the oscillators have warmed up and stabilized, adjust the frequency of the left channel oscillator to 168 Hz and that of the right channel to 318 Hz. At low level these two tones are quite discordant. Increase the level until the pitches of the 168-Hz and 318-Hz tones decrease to the 150-Hz–300-Hz octave relationship, which gives a pleasant sound. This illustrates the decrease of pitch at the lower frequencies. An interesting follow-up would be to devise a similar test to show that the pitch of higher frequency tones increases with sound level.

TheAshkOne - Vitality [A Trance Live Jam]

A Trance Jam Produced Using My Handmade Synthesizer

Paper Recommendation #1

0.04-mm2103-dB-A Dynamic Range Second-Order VCO-Based AudioΣΔADC in 0.13-μm CMOS

Abstract:

This paper presents a compact-area, low-power, highly digital analog-to-digital converter (ADC) for audio applications. The proposed converter is implemented using only oscillators and digital circuitry, without operational amplifiers nor other highly linear circuits. The ADC consists of two twin secondorder ΣA modulators, which can work both individually or in a pseudodifferential configuration. The proposed system has been implemented in a 0.13-μm standard CMOS technology. The single-ended configuration occupies an active area of 0.02 mm 2 , is powered at 1.8 V with a current consumption of 155 μA, and achieves an A-weighted dynamic range (DR) of 98 dB-A. The pseudodifferential configuration achieves 103 dB-A of A-weighted DR at the expense of doubling the area and power consumption.

Published in: IEEE Journal of Solid-State Circuits ( Volume: 53 , Issue: 6 , June 2018 )

DOI: 10.1109/JSSC.2018.2799938

Op Amps with Data Converters (Part 2)

It is useful to examine a few general trends in data converters, to better understand any associated op amp requirements. Converter performance is first and foremost; maintaining that performance in a system application is extremely important.
In low frequency measurement applications (10 Hz bandwidth signals or lower), sigma-delta ADCs with resolutions up to 24 bits are now quite common. These converters generally have automatic or factory calibration features to maintain required gain and offset accuracy. In higher frequency signal processing, ADCs must have wide dynamic range (low distortion and noise), high sampling frequencies, and generally excellent ac specifications.
In addition to sheer performance, other characteristics such as low power, single-supply operation, low cost, and small surface-mount packages also drive the data conversion market. These requirements result in application problems because of reduced signal swings, increased sensitivity to noise, and so forth. In addition, many data converters are now produced on low-cost foundry CMOS processes which generally make on-chip amplifier design more difficult and therefore less likely to be incorporated on-chip.
Analog input to a CMOS ADC is usually connected directly to a switched-capacitor sample-and-hold (SHA), which generates transient currents that must be buffered from the signal source. On the other hand, data converters fabricated on Bi-CMOS or bipolar processes are more likely to have internal buffering, but generally have higher cost and power than their CMOS counterparts. It should be clear by now that selecting an appropriate op amp for a data converter application is highly dependent on the particular converter under consideration. Generalizations are difficult, but some meaningful guidelines can be followed.

• Higher sampling rates, higher resolution, higher ac performance

• Single supply operation (e.g., 5V, 3V)

• Lower power

• Smaller input/output signal swings

• Maximize usage of low cost foundry CMOS processes

• Smaller packages

• Surface-mount technology

The most obvious requirement for a data converter buffer amplifier is that it not degrade the dc or ac performance of the converter. One might assume that a careful reading of the op amp datasheets would assist in the selection process: simply lay the data converter and the op amp datasheets side by side, and compare each critical performance specification. It is true that this method will provide some degree of success; however, in order to perform an accurate comparison, the op amp must be specified under the exact operating conditions required by the data converter application. Such factors as gain, gain setting resistor values, source impedance, output load, input and output signal amplitude, input and output common-mode (CM) level, power supply voltage, and so forth, all affect op amp performance.

Op Amps with Data Converters (Part 1)

Today we are going to talk about data conversion and associated signal conditioning circuitry involving the use of op amps.
The picture below shows a generalized sampled data system and some possible applications of op amps. The analog input signal is first buffered and filtered before it is applied to the analog-to-digital converter (ADC). The buffer may or may not be required, depending upon the input structure of the ADC. For example, some ADCs (such as switched capacitor) generate transient currents at their inputs due to the internal conversion architecture, and these currents must be isolated from the signal source. A suitable buffer amplifier provides a low impedance drive and absorbs these currents. In some cases, an op amp is required to provide the appropriate gain and offset to match the signal to the input range of the ADC.

Another key component in a sampled data system is the antialiasing filter which removes signals that fall outside the Nyquist bandwidth, fs/2. Normally this filter is a low-pass filter, but it can be a band-pass filter in certain undersampling applications. If the op amp buffer is required, it may be located before or after the filter, depending on system considerations. In fact, the filter itself may be an active one, in which case the buffering function can be performed by the actual output amplifier of the filter.

After the signal is buffered and filtered, it is applied to the ADC. The full-scale input voltage range of the ADC is generally determined by a voltage reference, VREF. Some ADCs have this function on chip, while others require an external reference. If an external reference is required, its output may require buffering using an appropriate op amp. The reference input to the ADC may be connected to an internal switched capacitor network, causing transient currents to be generated at that node (similar to the analog input of such converters). Some references may therefore require a buffer to isolate these transient currents from the actual reference output. Other references may have internal buffers that are sufficient, and no additional buffering is needed in those cases.

The output of the ADC is then processed digitally by an appropriate processor, shown in the diagram as a digital signal processor (DSP). DSPs are processors that are optimized to perform fast repetitive arithmetic, as required in digital filters or fast Fourier transform (FFT) algorithms. The DSP output then drives a digital-to-analog converter (DAC) which converts the digital signal back into an analog signal.

 Data converter amplifier applications:
• Gain setting
• Level-shifting
• Buffering ADC transients from signal source
• Buffering voltage reference outputs
• Buffering DAC outputs
• Active antialiasing filter before ADC
• Active anti-imaging filter after DAC

The DAC analog output must be filtered to remove the unwanted image frequencies caused by the sampling process, and further buffering may be required to provide the proper signal amplitude and offset. The output filter is generally placed between the DAC and the buffer amplifier, but their positions can be reversed in certain applications. It is also possible to combine the filtering and buffering function if an active filter is used to condition the DAC output.

LEVITATED MIAMI 2018

Various Artists

RELEASE DATE 2018-03-23
LABEL Levitated Music
CATALOG LEVC012
Listen : Beatport

THE BEST OF LEVITATED MUSIC 2016

Various Artists

2016 marks an extraordinary year for the trance scene and Levitated Music as well. The Best of Levitated Music 2016 includes 20 outstanding tracks by Manuel Rocca, Alex Wright, Mhammed El Alami, Derek Palmer, Aimoon, Madwave and many others, those which got massive support from mayor Djs such like Armin van Buuren, Aly & Fila, Paul van Dyk, Alex M.O.R.P.H, Sean Tyas and many others. Expect more stunning releases in 2017. Happy New Year trancefamily

RELEASE DATE 2016-12-26
LABEL Levitated Music
CATALOG LEVC009
Listen : Beatport

IBIZA CLOSING PARTIES 2016 - TRANCE

Various Artists

Ibiza Closing Parties 2016 - Trance is back by popular demand. tracks and remixes from LUCIEN, The Ashk One, Quervo, Costel Van Dein, Amitacek, Kayat feat. Clare Stagg, Quasi, Frainbreeze feat. Natune, Eddie Lung, Andrea Bertolini, Andrey Dobarin, Axis, RAMiNiO, Digital Fuzion, LITHH, Guido Hermans, Fong, Imida, Roman Messer & Mhammed El Alami With Julia Lav, Mhammed El Alami & Mobil, Kevin Crowley, Diher & R3dub, Alex V, Mino Safy, Marco Mc Neil, Kumar, Kanski

RELEASE DATE 2016-09-11
LABEL LW Recordings
CATALOG LWICP201604
Listen : Beatport

TRANCE HITS TOP 20 - 2016-06

Various Artists

A plethora of top-quality Trance awaits you and it's bound to leave you speechless. Following in the footsteps of all previous editions by harboring another twenty top-tier trancers, the 'Trance Hits Top 20 - 2016-06' is a temptation few are able to resist. With tunes such as the Sunny Lax Remix of Kyau & Albert - 'Made Of Sun', Alan Morris - 'Lost In Space', Stoneface & Terminal - 'Hypogean', and more, there's just no going wrong

RELEASE DATE 2016-06-24
LABEL Armada Music Bundles
CATALOG ARVA821
Listen : Beatport

LEVITATED, VOL. 1 MIXED BY MANUEL ROCCA

Various Artists

Anticipating our 50th release, we are thrilled to present you a brand new compilation series, 'LEVITATED' selected and mixed by the Venezuelan Trance Star Manuel Rocca. This first edition contains 11 wonderful tunes from Abora Music Label Group, featuring great artists like Illitheas, Miroslav Vrlik, Mhammed El Alami, Estigma, Alex Wright, Michael Retouch and many others. Pure and beautiful uplifting trance is what you can expect!

Get ready for this new journey that will elevate your emotions to another level. Enjoy

RELEASE DATE 2016-06-20
LABEL Levitated Music
CATALOG LEVC007
Listen : Beatport

UPLIFTING ONLY TOP 15: MAY 2016

Various Artists

Once again, Ori has personally hand-picked 15 of the very best tracks from the latest episodes of his Uplifting Only radioshow/podcast, and here they are!

The definitive show for orchestral uplifting trance, UpOnly airs on 38 radio stations around the world and has been named the #6 & #8 trance radio show in the 2015 & 2014 Trance Podium Awards, with 3 of its mixcomps reaching the #1 spot on the Beatport charts.

This edition of the monthly compilation features music from SoundLift, Afternova, illitheas, Mhammed El Alami, Plutian, Mike van Fabio, Type 41, InnerSync, Arrakeen, Plutian, Bernis, Andy Elliass, Johannes Fischer, and many more

RELEASE DATE 2016-05-19
LABEL Abora Music Compilations
CATALOG UOMC1605
Listen : Beatport

The Ashk One - Paradise

Early support from Armin van Buuren on #ASOT759

Here at Levitated always blows a summer breeze. The Ashk One arrives straight from Iran to make his debut on the label in the form of a 'Paradise', a stunning prog-lifter with rolling bassline, sweet vocal chops and catchy leads, all wrapped in a summery wave. Derek Palmer flips the original mix to create his banging uplifting interpretation
RELEASE DATE 2016-04-25
LABEL Levitated Music
CATALOG LEV047
Listen : Beatport

GREAT PERSIAN EDM, VOL. 1

Various Artists

Here is the another various artist bundle from Royal Comps, Great Persian EDM Vol. 1. Cutting straight to the chase, are 18 peak-time euphoric hits, hand picked and bundled into one DJ friendly album. Including tracks and remixes from All Persian Artists. With tracks that have been supported by many more big djs ... to name a few look no further for your fix of all things Chill Out , Dubstep , House ,Progressive and Uplifting.This is Great Persian EDM Vol.1 with Pure Persian Artist

RELEASE DATE 2015-04-13
LABEL Royal Comps
CATALOG RCMP08
Listen : Beatport

The Ashk One, AyNix - AUTUMN DREAMS

Tunisia & Iran are joining forces on Pulsar Recordings to bring you an exquisite trance track produced by The Ashk One & Aynix. Their debut single on our label called Autumn Dreams starts in a progressive manner, but soon starts revealing its melodic side of things by slowly introducing beautiful plucked instruments. Once you enter the breakdown, you will find yourself in a state of a dream just like the title suggested, welcoming you with those ethereal pads that soon transform into a big climax, driving you all the way to the end.

On remix front, we first have Miroslav Vrlik with whom many of trance listeners are already familiar with. His take starts with big uplifting beats and acid lines that pave the way for the beautiful and recognizable melodies from the original track, but this time with slightly more classical uplifting sound that we are sure many will find pleasing.

UDM provided for us a remix that starts in similar progressive take but with slightly more energetic sound than the reference track. The highlight of the breakdown here is an amazing piano melody that complemented so well those dreamy pads. When you think it can't get better the synths slowly creep in and take us away into an uplifting dreamland.

Last but not least is the remix from another upcoming producer from Poland, going by the alias of Sowa, which simply means owl in Slavic language group. His take is characterized exactly like the alias, as we all know that owls are the creatures of the night, so this track is no different. Featuring the mysterious pads in the beginning and late night atmosphere in the breakdown, with those amazing bell like sounds. The suspense starts building with rolling kicks until the climax comes and brings us light and joy that rounds it up for another great release

RELEASE DATE 2015-01-26
LABEL Pulsar Recordings
CATALOG PULSAR173
The Ashk One, AyNix, Miroslav Vrlik, UDM, Sowa
Listen : Beatport

BEST OF PROMIND RECORDINGS 2013

Various Artists

And the third year of first Persian trance label, Promind Recordings serve up the third part of their Trance Essentials collection series. Featuring 12 of the biggest tracks,in uplifting and progressive and vocal! all of which have picked up support from the worlds elite including Armin Van Burren , Above and Beyond , Aly and Fila,Sean Tyas, Manuel Le Saux, Rank1,Robert Gitelman and many many more Djs

RELEASE DATE 2014-01-16
LABEL Promind Recordings
CATALOG PMRCD08
Listen : Beatport

The Ashk One - OUT OF HORIZON

The first in our new Inov8 Sampler series is filled with three aural treats including music from Eric Strong & Malyar with U, M.D.K vs Christian Chatelle with Buenos Aires and The Ashk One with Out Of Horizon!! Three daring tracks rom three up and coming production teams

INOV8 SAMPLER 01

Eric Strong, MalYar, M.D.K, Christian Chatelle, The Ashk One
RELEASE DATE 2013-12-23
LABEL Inov8 Recordings
CATALOG INOV090
Listen : Beatport

AyNix - PLANETARIUM

Planetarium is the brand new release of the young and talented Aynix from Tunisia. This uplifting track will take you to the next level with the amazing melody, sound and the drums, which makes the track really perfect

Remixes :
AyNix, The Ashk One, Trizet, Francisco Echeverria, Silica
RELEASE DATE 2013-10-05
LABEL Kalimira Music
CATALOG MK036
Listen : Beatport

The Ashk One - Hear To Heart (Original Mix)

Once upon a time,Persian production The Ashk One is come with his next club destroyer. This time shifting to our main Promind imprint with the impressive 'Hear To Heart'.We kick off with that signature dirty bassline and funked up groove, subtly teasing you with glints of the melody as we progress. The break is a huge fanfare of gorgeous sounds, plucky piano keys, a big room melody and that raucous bass line which will cause maximum effect. Another essential cut once again from Ashkan, who has turned out a sure fire summer smash which his fans will love

RELEASE DATE 2013-08-14
LABEL Promind Dark Records
CATALOG PMDR014
Listen : Beatport

PROMIND IN PARIS CHAPTER III

Various Artists

Promind Presents the 'In' series where we are jetting off to some of the most exciting places on the planet every season and truly putting the Promind stamp on them. Ten exciting themed tracks from our past and present catalog! Our firstdestination is Tehran!13 Progressive,uplifting & Tech Trance numbers for dazzling lights of Tehran from the likes of; Proyal, Aron H, Amir Farhoodi, Denis Sender,Mike Oceanic,Eugene Karnak, Antonio Next,Shahead and Witness45 more

RELEASE DATE 2013-07-06
LABEL Promind Recordings
CATALOG PMRCD06
Listen : Beatport

StyleWriters - Limited (The Ashk One Remix)

Alien Tunes Recordings is proud to announce this additionnal release of the amazing progressive House tune From StyleWriters, ''Limited'' is as soulful and funky as ever, using some sexy disco samples, paired with the boys already trademark beats and production vibes.... So hot, we can already smell the summer coming! Not much more to be said, except the usual: LISTEN, DONWLOAD and PLAY IT OUT LOUD in the clubs and your

RELEASE DATE 2013-05-08
LABEL Alien Tunes Recordings
CATALOG ATR006
Listen : Beatport

Book Recommendation #1

One of the books which I came across reading it during my MA Thesis was ..........

The story begins since the day I got interested in the building blocks of ADCs and How they work, as it's a useful part in music production and audio recording and I've always had trouble with the limited number of ADCs in my audio interfaces, when I was trying to record big bands or orchestral.
most of the time I was opening two DAWs at the same time or was using some software to sync and use my multiple audio interfaces in one DAW.
So I decided to learn about it more and maybe try to build my own ADCs and add my decent Pre-amps to them to be able to capture multiple channels at the same time.
But as you may know, Data Conversion is something so unique in electronics so I decided to do some research and learn about it.
at the same time, I was about to finish my MA degree and it was time to give my thesis topic to my master in university.
So I decided to design and build an ADC as my MA thesis, thus I could solve my converter problem , finish my Thesis and also learn something.
One of the great books in this field which helped me to understand the whole field and find the way was this :
Data Conversion Handbook by Walt Kester

You can find almost anything about Data Conversion/Convertors like,

- Data Converter History

- Fundamentals of Sampled Data Systems

- Data Converter Architectures

- Data Converter Process Technology

- Testing Data Converters

- Interfacing to Data Converters

- Data Converter Support Circuits

- Data Converter Applications

- Hardware Design Techniques

more books about this topic coming soon.

Building A Passive RFID System.

Recently one of my university teachers asked me to work on an RFID project.
RFID Stands for Radio-frequency identification.
It uses electromagnetic fields to automatically identify and track tags attached to objects.

that was my first radio frequency project.
I've never worked with frequencies above 20khz as I'm always working on audio circuits.
so I did some research, asked few people and searched on the internet.
First of all, you have to know that mainly there is two kind of RFID systems.
1.Active 2.Passive
1. Active tags have a local power source (such as a battery) and may operate hundreds of meters from the RFID reader.

2. Passive tags collect energy from a nearby RFID reader's interrogating radio waves.
First, I decided to go for the active one as it's working with an external power source and it can work at long ranges.
All RFID systems contain receiver and tag.
I decided to build the receiver first and go for the tag after that.
as I didn't know anything about rf circuits I did some search on the internet for some active RFID receiver circuits and BOOM.
the active receiver circuits were really hard to build and the components were hard to find and forget about the tag part.

Plus, this kind of systems is getting used at big factories and mainly they are using it to transmit sensor data and .....
so I decided to go for the passive one because I'm a Genius.
There are 3 types of passive RFID systems.
LF Passive < 135 kHz
HF Passive  13.56 Mhz
UHF Passive  868 - 950 Mhz
(this one is active)
UHF Active  433 - 5.8 GHz

(you can find more data about differences in range and frequency at the table below.)

TheAshkOne - A Night Deep In Sky [A Lo-Fi Track]

Another Track produced with #handmade #synthesizers

TheAshkOne - Pink Afternoon [A Lo-Fi Track]

Here is a Track of mine produced with my #Handmade #synthesizers.

Hello World!

Hello World!