امروز صبح سایت http://www.ypsilonelectronics.com/technology.htm رو چک میکردم که مطالبی نوشته بود در مورد طراحی و بنظرم این کمپانی یونانی باشه .

طراح این شرکت روی طراحی ترنسفورمر ایده هایی داره که جالبه و برخلاف 90 درصد برندها که از ترانس توروید با سیم پیچی مسی استفاده میکنند این برند از ترنسفورمرهای double C core استفاده میکنه.

این طراح آمپلی فایری ساخته تو مد سینگل اندد که با موازی سازی ماسفت 100 وات Class A Single ended میگیره و مثل نلسون هم به سادگی (کمتر از 3 طبقه مدار) معتقده و هم به استفاده از ماسفت در طبقه خروجی. میدونیم منحنی مشخصه ماسفت ها نسبت به ترانزیستور های بایپولار حساسیت کمتری به تغییر دما دارند و مشخصه خروجی شون (برخلاف ترانزیستور های بایپولار که سورس جریان هستند) مثل لامپ سورس ولتاژ هستند.

البته این طراح تعصبی روی لامپ یا ماسفت نداره و مثلا در طبقه اول و یا در رکتی فایرهاش از لامپ استفاده میکنه و نظرش اینه توپولوژی مهمتر از المان هست.

من ایده ای رو در مورد ماکرو و میکرو در بخش measurement یک دستگاه داشتم که بحثش رو ادامه خواهم داد و باید بگم پری آمپلی فایر این شرکت عین Lamm یکی از بهترین پاسخ ها رو از دید من در measurement ها داره.

یک مصاحبه هم با آقای Demetris Backlavas طراح این شرکت اینجاست که پیشنهاد میکنم حتما بخونید :

http://www.theaudiobeat.com/visits/ypsilon_interview.htm

مطالب زیر در سایت خود برند هست که اینجا آوردمش :

TECHNOLOGY

It took over 10 years of dedicated research in which time we explored the world of quality amplification. Our goal was to reach the absolute limits of excellence. In order to achieve this, one has to evaluate and experience every available technology in amplification circuits and components. Our experience to true live music gave us the opportunity to have a reference. Combined with our background in electronics and sound engineering we were able to explore what is possible in high quality reproduction.

There are two schools of how an audio signal can be amplified depending on the active components used: Solid State and Vacuum Tubes.

We came to the conclusion that the real dilemma is not Solid State versus Vacuum Tubes but Single-ended amplification versus Push-Pull amplification.

PUSH PULL

In Push-Pull class A or class AB two (or bank of parallel) active components are used, where one sinks current and the other sources current. In class AB operation the problem is that crossover distortion produces a cold and harsh sound. In class A operation most of the times the two halves are not the same components (PNP with NPN transistor or P-channel with N-channel mosfet). In quasi-complementary topology where two same components are used in the two halves, the problem arises from the different topology e.g. One half NPN transistor common emitter, the other half common collector. To minimize distortion various topologies have been used with different types of feedback. e.g. voltage feedback, current feedback, nested feedback/error correction, leading down a one way root to lifeless music. Even when there are two same halves in a Class A pentode or a triode P-P amplifier the sound is not as convincingly natural. This happens because one half acts somewhat as an “active” current source to the other half and thus loading each other producing a mechanical sound. An additional problem is the phase-splliter stage. There is no way it can be done in a consistent manner with active devices. The conclusion is that P-P is not the way forward for reaching the best.

SINGLE-ENDED

In Single-Ended amplification only one (or bank of parallel) active component is used. This demands operation in Class A, where current flows independently of the audio signal. Generally single-ended amplifiers are low wattage tube amplifiers. They provide musical involvement when realized properly. Most commonly used big output tubes are 211,845 and 833. The drawback is that in order to achieve maximum available power they have to be driven in class A2 (Grid starts to draw current from the previous stage). The result is a difficult and awkward load for the driver stage that starts loosing its consistency. E.g. 211 in pure class A delivers about 12 watts, after this and up to 25-30 watts starts to draw up to 30-50mA. The load that the driver stage sees is not constant during the full sinewave. Paralleling multiple tubes, also, is not a solution. Each tube loads the others in a strange way due to differences between each other. This causes a harsh and edgy sound. Also the measured distortion contains more odd harmonics (3th,5th,7th). Another issue that needs to be considered is the output impedance. Without feedback this is normally more than 1,5-2 ohms. The amplifier will alter its frequency response in loudspeakers with big dips and peaks in their impedance curve changing the tonal accuracy of the loudspeaker. A loudspeaker with a very even impedance curve should be used with such amplifiers.

THE SOLUTION

Single-ended amplification provides something that no P-P could ever provide. It is closer to the “real thing”, music flows in a way that happens only in live unamplified performances. By incorporating a unique single-ended mosfet output stage on the SET-100 we achieved on having all the virtues of a big single-ended triode output stage without having its drawbacks. We manage to have more power and drive with transparency, musical involvement and above all with music flowing naturally. With only two gain stages, tube input with tube rectification and mosfet output, without using overall feedback we achieved on having output power more than 100 watts, enough gain, and sufficient output impedance. In the SET-100VS amplifier which is a 2X 40Watt rms stereo amplifier we chose the GM-70 output tube. It is a linear ragged direct heat triode able to deliver high power for a S-E design, with a sound that is very musical and powerful at the same time. By driving the tube with a interstage transformer coupled driver stage, with matched characteristics to the output stage, the distortion is kept low for a no-feedback design. The outcome is an all valve S-E-T amplifier that provides a tonal palette of immense width and produces music in its true natural scale, but above all it brings you as close as possible to the music event, to feel the music rather than hear it, to be touched and overwhelmed by the deeper feeling, with music emerging and not only sounding in a clear undistorted way.

DIGITAL

In the first CD players presented to the market in the 1980’s, analogue filters were used at 22 kHz to reject out of band images of the audio signal. These filters cause a big phase shift in the audio spectrum and a slow response to transient signals. They are also very expensive to implement, so the stop band attenuation was applied in the digital domain with so called digital filters. In digital to analogue conversion these are interpolation filters, whereas in analogue to digital conversion filtering is done with decimation filters. With interpolation filters, data is added mathematically and calculated from the originally retrieved data of 16/44 kHz. This process is called “oversampling”. The result is that the sampling frequency of 44khz is increased to 96khz or 192khz and cheaper and more effective analogue filters can be used to reject the out of band noise. Even when this process is performed by powerful DSP (Digital Signal Processing) devices the end result is never like the originally retrieved data. Tremendous accuracy is required to retrieve clock (master, bit) data in order to keep jitter levels low. With interpolation filters, music sounds more processed and clinical. To avoid the problem of the high accuracy requirement of retrieved master clock data, a technique was used called “upsampling”. The data in the DAC input are interpolated and re-clocked by a local clock generator, thus achieving low jitter since the clock generator itself is within a short distance of the DAC. Then data are sent to the digital filter and the upsampling is done again with interpolation. By not using interpolation filters, the sound is more natural, the image depth and dimensions are open and better defined. Also, analogue filters colour the sound and affect dynamics in a negative way. Of course, by not using analogue filters, the DAC’s measurements include the out of band noise. But, it sounds more open, with bigger scale and a more analogue-like presentation. Remember that the basic idea behind DSD technology was to get rid of the digital filters used in PCM. Unfortunately the industry did not embrace it but instead, kept interpolation in one way or another. With the Ypsilon CDT-100 and DAC-100, neither oversampling nor upsampling are used. A very linear and accurate chipset are implemented. I/V conversion is accomplished by a specially designed transformer, designed and built in-house. The analogue stage of the DAC1-00 is a single ended class triode transformer coupled at the output. The power supply uses a valve rectifier and choke regulation. All signal and power supply transformers are designed and manufactured by YPSILON ELECTRONICS.

By using only the best materials available in the DAC-100 and combining it with CDT-100 the sound can only be compared with the best analogue sources. You will be astonished!!

لیست تحلیل ها :

http://www.stereophile.com/content/ypsilon-pst-100-mkii-line-preamplifier

http://www.dagogo.com/View-Article.asp?hArticle=95

بد فکری نیست یک انجمن عاشقان برف و بارون بزنیم همگی بریم زیر برف و بارون پیاده روی، خیلی باحال شده این هوا.

آقای بهرام عزیزی رو حتما بیشتر دوستان میشناسند و از نزدیک با این شخصیت آروم ، متین ، دوست داشتنی و فوق العاده بی ادعا آشنا هستند .

آقای عزیزی رو من دیروز دیدم و البته این اولین برخوردم با ایشون نبود. آقای عزیزی متخصص تعمیرات و مودیفای سیستم های الکترونیکی هستند که شامل سورس و پری و پاور میشه.

ایشون سالهای خیلی زیادی هست که در این کار تجربه دارند و برخی از واردکنندگان در ایران محصولاتشون رو برای تعمیر و سرویس به دست ایشون میسپارند.

من بیش از تخصصشون برام نوع شخصیتشون جالب بود، خیلی خوب برخورد کردند و از من به گرمی پذیرایی کردند.

امیدوارم موفق باشند و ازشون بخاطر این مصاحبه تشکر میکنم.

شماره تماس ایشون 88832044 هست و شما اگر نیاز به تعمیر دستگاهتون داشتید میتونید با ایشون تماس بگیرید.

تعدادی Dealer تو دنیا هستند که برندهای خوبی رو عرضه میکنند، یکی از اونها Soundscape HiFi هست که تو زمینه آمپ های کم وات لامپی و بلندگوهای با حساسیت بالا فعالیت میکنه.

http://www.soundscapehifi.com/

برندهای زیر رو این فروشنده میاره :

Living Voice
Lavardin Technologies
Artemis Labs
Jan Allaerts
Isenberg Audio
Da Vinci Audio Labs
Kondo Audio Note Japan
Horning Hybrid
Audolici
Tron Electric
My Sonic Lab
KAB
Neodio
TS Audio

سایت مقابل هم سیستم مشتریان این فروشنده رو نشون میده : http://www.soundscapehifi.com/client-systems/index.php

قصدم از این مطلب بیشتر از معرفی Soundscape معرفی بلندگوساز انگلیسی Living Voice بود که بلندگوهای درایوری 94 دی بی اون رو تو خیلی از نمایشگاه ها با Audio Note kondo ژاپن دمو کردند و این بلندگو میتونه گزینه خوبی برای کسانی که kondo دارند باشه.

روی عکس بالا کلیک کنید تا ابعاد بزرگتر رو ببینید.

من قبلا در مورد سورس دیجیتال Playback Designs صحبت کردم ، اتفاقی مصاحبه ای رو از طراح این برند دیدم که جالب دیدم اینجا بیارمش.

http://www.positive-feedback.com/Issue41/ca_koch.htm

http://www.ultrahighendforum.com/viewtopic.php?f=200&t=642

Exclusive Interview with Andreas Koch of Playback Designs
August 7, 2010
By: Frank Berrryman

Andreas Koch Of Playback Designs, whose MPS-5 CD/SACD player and MPD-5 digital-to-analog converter have received universal acclaim, was kind enough to consent to an exclusive interview with the Ultra High-End Audio and Home Theater Forum.

Thank you for agreeing to be interviewed for the Ultra High-End Audio and Home Theater Forum. Most of our members are unfamiliar with your background. Would you give us a brief biographical sketch?

First of all, thank you for the opportunity to talk about some of the technologies I have been researching and developing over the past 29 years. It is my hope that these technologies continue to contribute to the ever evolving and incremental improvements in sonic performance that we can experience in our industry.

My field of experience has always been digital signal processing, and because I already pushed the limits of the technology while studying it at the University in Switzerland where I built the fastest signal processor, I quickly became involved in digital audio right at its beginning, which then seemed so fast that hardly any hardware could keep up with it. At Studer ReVox I researched the theory and built the hardware for the first fully asynchronous sample rate converter with completely arbitrary frequencies. It was 1982 and barely a 16-bit world. I recognized that 16 bit would not be good enough, so I expanded my hardware to 24 bits.

Shortly after that I joined Dolby Labs in San Francisco as its first DSP engineer. I was surrounded by all these brilliant analog engineers and together we pushed the technical envelope and built the AC-1 audio compression scheme for TV broadcast. That was just the precursor of what I added shortly afterwards, the first real time processor to implement algorithms that later evolved into the widely used AC-3 compression which as many of you know is what DVD-Video and movie theaters are based on.

Meanwhile in 1987 computer technology had gotten just fast enough to handle the most basic tasks in digital audio. It was a new frontier calling me and I decided to return to Studer to head a project where digital audio would be recorded onto computer hard disk for editing. Soon I launched one of the most complete computer audio platforms for professional recording, editing and mixing, known as Dyaxis.

What could possibly be the next technical frontier after such a gigantic project? Sony helped me answer that question: launch a new consumer audio format with more channels and vastly improved sonic performance. While developing the very first SACD mastering recorder / editor (“Sonoma”) with native DSD processing, I also had to develop a new kind of A/D and D/A converter that would be able to highlight the advantages of SACD. Price was not the objective. It was only sonic performance that mattered. Sonoma is still used today to master many SACD’s.

Following my work with Sony, I had gained so much experience and knowledge of technologies in digital audio processing that I thought I should try marketing my know-how as an independent contract engineer. I was very successful with it up until very recently when Jonathan Tinn and I teamed up and formed Playback Designs.

What led you to start Playback Designs?

Jonathan and I met when we were both heavily involved with EMM Labs. Once both of us left EMM Labs, we kept in touch as friends and started talking about forming a new company that would allow us to implement our own ideas without the limitations put on us by past or existing products or any corporate culture. The decision was made and we formed Playback Designs. I started with a clean sheet of paper and close to 30 years of experience and designed the Playback Designs MPS-5. This player represents the pinnacle of digital audio design. There has never been another player like it and is what I believe to be the true “state of the art”.

What in general is your design philosophy?

In audio design as with almost anything else, keeping it simple is almost always the hardest and most beneficial thing to do. When designing, I keep this in mind at all times. I am constantly considering how to simplify or shorten the signal path or simplify an algorithm. This allows a purer and more lifelike sound which I believe is truly the goal.

What is a “two-dimensional” DAC?

This is discussed quite in-depth on our website. Perhaps it would be best if I repeated it here:

2 Dimensional DAC Technology and Computer Audio

“Audio is represented in a y/x-axis system: the y-axis for amplitude and the x-axis for time. Mostly because of analog audio’s sensitivity problems in the y-axis, digital audio was introduced. But digital audio not only quantizes the y-axis, it does so as well on the x-axis. Sounds like we got more than we wanted – true and too bad. A typical state-of-the-art DAC converts between quantization levels in the digital y-axis and the analog y-axis and is completely transparent and open as to what happens on the x-axis (time domain). Sounds like we forgot the quantization on the x-axis.

This oversight forced us to treat digital audio signals as if they were analog: use special cables, use all kinds of mechanical devices for our CD players, power conditioners for digital audio etc. Looks like we just shifted the original problem from the y- axis to the x-axis, but the issues are still the same. Instead of interference or crosstalk we now call it clock jitter.

Almost all DACs available today deal with the y-axis only and rely on external devices for the x-axis, such as complicated master/slave clock arrangements or external sync clock generators. At best these devices are band-aids on a wide open wound deep inside the DAC. They help, but do not resolve the problem at the source. We need a 2-dimensional DAC that not only works on the y-axis, but also on the x-axis. With this we can separate the digital world completely from the analog one and render any digital cable, transmission format, storage media and application completely irrelevant to the final sonic performance. The only analog problem that we still have then is the separation of the power supplies for digital and analog.

The DAC inside the Playback Designs product line does exactly that: clock jitter from incoming digital audio signals can be described as an analog signal that gets mixed together with a quantized digital signal (our ideal and constant sample rate clock). So before any processing can happen we need to bring these 2 components into the same domain: The Playback Designs system quantizes the clock jitter into a digital signal, where it then can be subtracted from the original sample rate while the latter is converted to analog at the same time. Of the course, the DAC also works independently in the y-axis by using a set of unique algorithms in a completely discrete architecture (not even a single Op-Amp is used).

Tests have shown that the DAC inside the Playback Designs product line can be fed by any digital source including a PC, an inexpensive Discman, a DVD player, or high-end CD transport and none of them seem to make a difference on the sonic performance of the analog output signal. Ultimately this means that as long as you are sending our DAC truthful complete bits the source does not make a difference. We believe if you own a home computer, you already own a music server that cannot be sonically bettered!”

Would you explain what DFAS (Playback Designs Frequency Arrival System) is, how it differs from other jitter reduction strategies, and how it not just reduces jitter but completely eliminates it?

Jitter is real and it exists – some of it is good, some bad. But rather than trying to tell the difference in your incoming signal and trying to filter it, I approach it from a different angle.

If you tell your friend to follow you in your car, he will inevitably regulate the speed of his car up and down to stay in sight of your car. No matter how constant you drive your car your friend will vary his speed in small amounts. That is how a PLL (phase locked loop) works and is commonly used to generate an audio clock in a DAC.

Imagine you tell your friend where you intend to go and what time exactly you expect him to be there. That knowledge enables your friend to travel completely detached and independently from you by calculating a constant speed that brings him to the destination at the exact time. Life really gets easy when we communicate and can trust each other, we just need to create a similar environment in audio where that is possible and that is what I did with PDFAS.

This is a brief analogy of what I am doing and although I realize I did not answer your question with real technical detail, the reason is simple. This proprietary knowledge is one of many things that separate us from our competitors. If I share this knowledge here, other companies might implement my discoveries in their products and we might lose some of our uniqueness.

In your judgment, at what level does jitter become audible, and what should audiophiles listen for to determine whether jitter is present in their systems?

There are so many different kinds of jitter and many people have already tried to characterize them – and that is fine, but there isn’t a simple criteria at which point you can determine that it is audible or not. Even if you eliminate the jitter in your DAC you can still hear jitter from the A/D converter that was used in the recording, because it was embedded in the digital files (worst of all) with no chance of fixing it later. Also, for various reasons it might be desirable to still have some jitter in the DAC, completely uncorrelated to the audio signal. Where do you draw the line? Everybody does it differently.

I take it you do not rely on third-party DAC chips in your components. Would you explain how you implement digital to analog conversion?

While not revealing any trade secrets I still use the same very basic concepts used in most chip solutions, but by using discrete components and implementing all the algorithms myself I can control every single step of the process. For instance, in one algorithm I found that the commonly used 32 bits or 56 bits were not sufficient. I used an insanely higher number of bits just because I could hear a difference. Discrete steps also allow me to introduce novel algorithms that address some of the issues of common signal processing. That is where the sky is the limit and I can provide improvements mostly in the form of software upgrades, all thanks to a flexible and discrete platform.

The MPS-5 plays back both CDs and SACDs. Obviously they employ different digital architectures – PCM and DSD. Is the digital to analog conversion handled by two separate circuits or do you, for example, convert DSD to PCM prior to processing?

This is really a funny question because taking a DSD signal and converting it to PCM prior to processing is probably the worst thing you could do and I realize that there are a number of companies currently doing this very thing. I would liken it to taking a high resolution photograph and sizing it down. When you view the sized down version, it still looks great, just smaller. Now try enlarging it to its original size and all of a sudden the picture becomes extremely blurry. It is the same as converting DSD to PCM and then feeding it to a D/A which always has a stage of some form of DSD before converting the signal to analog. Very simply, the most direct route between two points is a straight line. Why would you want to take a different route if it was unnecessary?

If I am reading your website correctly, you “oversample” the data stream 128 times during processing. What is oversampling and how is it beneficial?

Yes, Playback Designs’ products take all data, whether DSD or PCM, and oversample it to 128 times the fundamental frequency (44.1 or 48kHz). If you oversample a digital signal by an infinite amount it automatically becomes analog. You want to oversample your digital signal as high as possible in order to get as close as possible to “analog” before the DAC actually outputs analog. There is nothing new about this, every commercially available converter chip does this internally to some degree.

In addition to oversampling, do the MPS-5 and MPD-5 upsample? If not, why not? If so, what are the sonic advantages of upsampling?

Although there are academic differences to be argued, there really is no difference between oversampling and upsampling. To me, it all means the same thing.

I understand that the MPS-5 can be expanded to a full six-channel setup or a four-channel system with center channel mixdown to the front left/right channels. Does this mean that in its six-channel configuration the MPS-5 can play back multi-channel SACDs? How do you implement center channel mixdown with the four channel option?

The MPS-5 can be programmed to read the multichannel area of SACD’s. Therefore, it can generate 6 channels of audio. Its internal DAC converts the front left and right channels and each additional MPD-5 that you use would convert the next pair until all 6 channels are converted. In the case of a 4-channel only setup, the MPS-5 can be programmed to mix the center channel into the front left and right channels. The LFE channel usually contains very similar information and it can be derived from either the front left or right channel directly. The mix down process in the MPS-5 is done in the DSD domain without detour to PCM. The one additional MPD-5 required then converts the surround left and right channels.

Both the MPS-5 player and the MPD-5 digital to analog converter have AES, S/PDIF, Toslink, USB and PlayLink inputs. Several [b]Questions: first, the USB input is limited to 16-bit 44.1kHz/48kHz signals. Is that a limitation of the USB interface? What USB to AES or S/PDIF converter would you recommend for computer playback of high resolution digital files, including playback from a laptop? Finally, what is PlayLink and do you have any plans for implementing it in the near future?[/b]

The USB input is primarily targeted for applications where CD libraries are stored on personal computers and played directly via the USB connection. Of course, the USB interface in general can implement much higher data rates. The MPS-5 can be upgraded once the corresponding implementation becomes available.

The conversion between digital formats such as USB to S/PDIF or others should be irrelevant as long as you don’t lose any data bits. It is the converter’s responsibility to remember the x-dimension as explained earlier in question #4. But since most converters do not address the x-dimension, I can see that your question can be important to some people.

Frankly speaking, we feel the best all around way to connect a computer to one of our DACs, especially for high resolution file playback, is with a sound card like a Lynx AES 16. Although I do not know what one can use for a laptop, there might be a similar product that would work equally as well. The AES connection is excellent and long cable runs are used quite often with great success. The same cannot be said for USB.

Playlink is already implemented and used to link the MPS-5 to its companion MPD-5. The format is flexible enough to expand to all kinds of other applications in the future.

What special lengths have you gone to assure that the analog output stage in the MPS-5 and MPD-5 meets the same high quality standards as the digital to analog conversion stage?

The analog output stage used in the Playback Designs converter is designed with the same exact goals and principle as its digital front end: keep it simple. We use the highest quality, best sounding parts available, along with the shortest possible signal path. The design breaks quite a few rules that have been assumed so naturally by many other designers. By always looking at new ways of simplifying our design, we seem to also simplify the sound.

In closing, is there anything else you would like to tell us about your company and products?

On the very opening page of our website, we state: “Playback Designs imagines, creates and manufactures the highest performance jitter free digital playback systems available for the most discerning of listeners.” We truly believe in this and will continue to live up to this statement in the future.

Thank you again for taking time to speak with us today.

من وبلاگ نویسی رو با وردپرس شروع کردم و هنوزم به وردپرس علاقه مندم هرچند جوملا امکاناتش بیشتره و اونو از وبلاگ به سایت نزدیک میکنه اما من هنوز ترجیح میدم با وردپرس کار کنم.

وردپرس هم خیلی امنیت خوبی داره و هم برای کار راحته و امکاناتش کاملا برای وبلاگ نویسی راضی کننده هست در ضمن سئو خوبی هم داره .

من مطمئنا حوصله نصب و کانفیگ وردپرس برای کسانی که علاقه مندند در مورد های فای وبلاگ داشته باشند با دامین اختصاصی رو نخواهم داشت اما به این دوستان خصوصا آقای عارف (آقای عارف رو با ساخت آمپلی فایر 211 بنام ستسون قبلا معرفی کردم) پیشنهاد میکنم سایت زیر رو ببینند و نحوه نصب و راه اندازی وردپرس رو یاد بگیرند چون حیف هست کسی مطلب بنویسه در مورد های فای و این مطالب روی سرور مثلا بلاگفا باشه. نمیگم بلاگفا امن نیست اما داشتن دامین و هاست و نصب وردپرس بنظر من برای کسی که در درازمدت قصد اطلاع رسانی داره گزینه بهتری هست.

هزینه سالیانه دومین و هاست چیز خیلی کمی هست و وردپرس هم که رایگان هست.

جادی مطلبی رو در مورد آموزس وردپرس نوشت که من بفکر افتادم این مطلب رو بنویسم.

سایتی که آموزش وردپرس رو میتونید به رایگان دانلود کنید اینجاست :

http://www.ebusinessfa.com/?p=1396

منم لازم دیدم از کسی که زحمت نوشتن این کتاب رو کشیده تشکر کنم و همچنین از تیم وردپرس فارسی.

موضوع فهم و یادگیری در انسان موضوع خیلی پیچیده ای برای مطالعه هست اما من به اختصار در مورد نگاه خودم مینویسم که شاید کاملا غلط باشه.

بنظر من فهم دقیق یک موضوع یا گزاره یک فرایند هست و نه یک چیز یک دفعه ای که از خوندن یا شنیدن حاصل بشه.

گزاره ها و به بیانی کلی تر زبان نقش زیادی در تسهیل درک و ارتباط داره اما خود درک و فهم یک فرایند ذهنی است که گزاره های بیرون میتونند تریگر باشند برای شکل گیری این تغییر. غیر از این حالت میشه به خاطر سپاری و نه چیزی بیشتر.

به عبارتی قدم اول درک به خاطر سپاری گزاره هست اما قدم های زیادی تا انتها داریم و برای همینه تعداد کسانی که چیزی رو دقیق فهمیدند تعدادشون خیلی کمتر از تعداد کسانی که اون چیز رو به خاطر سپردند هست.

خیلی ها بعنوان استاد دانشگاه نظریه انیشتن رو درس میدند اما واقعا درک کمی از جریان داشتند اما عده معدودی هم هستند که این فاصله به خاطر سپردن رو تا درک کامل طی کردند.

درک مفاهیم صدا هم بدون تجربه ممکن نیست. بدون درک عمیق فقط در سطح میتونیم با هم گفتگو کنیم اما ارتباط عمیق تر با درک عمیق تر شکل میگیره.

تو حوزه صدا ما نیاز به تجربه بیشتر و شرایط بهتر تست برای درک عمیق تر مفاهیم داریم و نه بحث های بی نتیجه سطحی.

فرض کنید یکی بیاد همه چیزها رو در های فای دقیق و درست توضیح بده و فرض هم کنیم همه این توضیحات 100 درصد درست باشه اما بنظر من چیز خیلی زیادی به ما اضافه نمیشه چون درک صدا فرایندی هست که ما باید در خودمون شکل بدیم و با شنیدن و تست و تجربه آگاهی هامون رو گسترش بدیم. اطلاعات درست بیرون فقط کمک میکنه که ما سریعتر این تجربه رو طی کنیم و به مرحله آخر برسیم.

دنیای صدا از نگاه من 90 درصدش تجربه شنیداری در شرایط مناسب هست و 10 درصدش مطالعه نظرات دیگران.

خوب بشنوید صداها رو و به خودتون و تجربه هاتون متکی باشید. بدون پیش قضاوت تجربه کنید و نتیجه رو خودتون بگیرید و نگذارید ایده های دیگران بجای کمک به شما مانعی برای درک درست صدا باشه.

همه چیز به خودتون بر میگرده پس تنها راه حرکت کردن تجربه کردن بیشتر هست.

http://en.wikipedia.org/wiki/DIY_ethic

http://en.wikipedia.org/wiki/DIY_culture

http://en.wikipedia.org/wiki/DIY_audio

مفهوم DIY بنظرم نیاز به توضیح روشن تری داره چون برخی در ایران تصور میکنند DIY یعنی چیزی که یکی با دست ساخته و قصد داره به بقیه بفروشه.

تفکر DIY ساختار مشخصی داره و اولیش اینه که هر کسی خودش آزاد باشه تا با فراگیری اطلاعات اقدام به پیاده سازی ایده هاش بکنه و از تجربیات دیگرانی که تو این حوزه فعالند استفاده بکنه. بنابراین هر علاقه مند به DIY در مسیر حرکتش هم اطلاعات میگیره و هم از تجربیاتش دیگران میتونند استفاده کنند و من تا حالا ندیدم کسی خودش رو در ابتدا عاشق  DIY معرفی کنه اما هم اطلاعاتش رو بخواد پنهان کنه و هم چیزی که ساخته رو بخواد به کسی بفروشه.

فروشندگی چیز بدی نیست اتفاقا خیلی هم خوبه اما ربطی به DIY نداره. حالا چه برند بفروشیم و چه بلندگویی که خودمون ساختیم رو بفروشیم و چه خازن و ترانس و لامپ بفروشیم در هر صورت یک فروشنده هستیم.

DIY یک نوع تفکر هست که هدف هایی داره :

1. پول اضافه ندهیم اگر میتونیم همون رو خودمون بسازیم

2. به بیرون وابستگی کمتری داشته باشیم و دقیقا چیزی رو که لازم داریم بسازیم

3. نتیجه تحقق ایده هامون رو در دنیای واقعی ببینیم و لذت ببریم

4. هم یاد بگیریم و هم به جریان آزاد اطلاعات درست کمک کنیم تا دیگران بعد از ما سریع تر و بهتر به نتایجی که دوست دارند برسند

 Ellen Lupton embellishes these thoughts in her book D.I.Y. Design It Yourself:
“Around the world, people are making things themselves in order to save money, to customize goods to suit their exact needs and interests, and to feel less dependent on the corporations that manufacture and distribute most of the products and media we consume. On top of these practical and political motivations is the pleasure that comes from developing an idea, making it physically real, and sharing it with other people.“

علاقه مندان به DIY منطقا به همدیگر توهین نمیکنند ، تعصب احمقانه ندارند ، خودشون رو رقیب هم تلقی نمیکنند ، به هم حسودی نمیکنند و سعی در بالاتر جلوه دادن خودشون نمیکنند چون منافعی از این کار ندارند بلکه به هم کمک میکنند اما فروشندگان بخاطر منافعشون ممکنه سعی در خراب کردن رقیباشون بکنند. فروشندگان در سراسر دنیا ممکنه به چرت و پرت گفتن و اطلاعات غلط دادن هم روی بیارند برای فروش اجناسشون اما علاقه مندان به DIY خودشون رو به نوشتن چرت و پرت، تعصب و تبلیغ روی یک مورد خاص و دادن اطلاعات غلط و مورونی مشغول نمی کنند.

اگر به وبلاگ یا سایت کسانی که در خارج به DIY علاقه مندند سر بزنید میبینید مثلا همین رومی خیلی دقیق هر چی ساخته رو شرح میده و کمک میکنه مخاطب به نتیجه برسه.

با اینکه رومی یکی از ارزشمندترین منابع اطلاعاتی رو اینترنت هست من تا این لحظه ندیدم اطلاعاتی رو بخواد از کسی پنهان کنه و کلا تو فرهنگ DIY ما علاقه مند به به اشتراک گذاشتن اطلاعات هستیم و نه پنهان کردن اطلاعات برای فروش چیزی.

همیشه به نتایج فکر کنید مطمئنا رویکرد بهتری خواهید شد.

خوش بگذره

خبر جدید اینکه من با بلندگوی Magneplanar 3.6 دارم خداحافظی میکنم ، متاسفانه بدلیل فروش کم سیب ها در چند ماه اخیر و نداشتن نقدینگی کافی از خرید این بلندگو منصرف شدم.

الان هم بدلیل مشکلات مالی هم سورس و هم آمپ رو گذاشتم برای فروش و کلا دارم بیخیال داشتن ست آئودیو میشم.

اینم سرگذشت های فای ما …

یک مقاله از اندی طراح آئودیو نت آنکورو بخونید جالبه هر چند کمی از نوشته هاش آبکی هست:

http://www.drtube.com/schematics/an/sp12-ankoru.pdf

Musical information is a dynamic four dimensional continuum, like the one posed by the relativity theory, consisting of three spatial dimensions and time, all inseparably inter-related. A system for music recording and reproduction must transfer this continuum and faithfully reconstruct the original sonic performance in the listening room. Test instruments are supposed to ensure that the transformation of the continuum is linear and accurate to certain parameters.
Alas, these instruments and the mathematical models that we use, such as the relativity theory, lack the spontaneity and emotional content vital to music. Somehow our primitive method of recording scratches into a vinyl disc captures some of this emotion, and the lump of rock we call a stylus is able to extract the information and convert the vibes into a signal ready for the amplification chain. The amplifier, therefore, must not only perform well electrically, it must also convey emotion in order to fully satisfy both the analytical mind and the inspirational soul.
These days, science is beginning to discover an essential, almost mystical, interconnectedness of everything. It is intuitively obvious that the character of the universe on a macroscopic scale should rely on the properties of the subatomic particles of which it consists. At the same time, the character and properties of those particles is defined by the universe at large, the whole system mysteriously holding itself up by its own bootstraps, each piece of the giant jigsaw fitting exactly into place without deficiencies or excesses.
It is only by virtue of an intellectual gesture that we perceive a condensed, solid, and definable part of the web of reality, yet we have deceived ourselves into thinking that our mental creation is the be-all and end-all of existence.
Most of our old scientific “laws” – including those currently used to judge sonic performance – are only close-ups of the whole picture. I’m afraid we are not seeing the wood for the trees.
Certainly these measured parameters do have some relevance in terms of overall performance, but to recreate a musical event, an amplifier must work on both a macroscopic scale as a part of a communication system between the performance and listener, and on a microscopic level as a collection of valves and parts which must be tamed and optimized for the task at hand.
A magazine article can only skim the surface of any design philosophy and, of course, there will be shouts of “what the *@$! is this guy on?”, but I hope my discussion of the Ankoru design will be interesting nonetheless.
Starting with the basic precept that each part of the amplifier should fit exactly into place, and have a character defined by the overall requirements of the system, the validity of feedback and push-pull operation, two pillars of traditional amplifier design, are immediately called into question. These concepts are purely intellectual constructions, created in laboratories with no motivation from natural music, and I am convinced that they detract from sound quality as a result.
In practice, the ultimate purpose of feedback and push-pull operation is to make amplifiers easier to make not better. In any event, reducing harmonic distortion to vanishingly small levels and increasing bandwidth from DC to cosmic rays does not make a more musically satisfying amplifier. Specs must give some satisfaction though, ‘cos we all know a guy who slinks off to the bathroom with a copy of his tranny amp spec sheet!
I agree that limitations such as distortion and bandwidth abhorations unquestionably colour the sound and should be eliminated, but beyond that I maintain that there are more important areas to be considered if musicality is the ultimate goal.
According to my way of thinking, all of the above leads to the assertion that the overall topology of an amplifier must be single-ended and there must be no feedback. Transistors and all things silicon sound unnatural … put sand in the signal path and you get gritty sound! So, let’s proceed directly to valves and, in particular, the simplest and purest amplifying device.
The materials used for the construction of the passive elements of the amplifier are just as important since the signal must pass through them. Every material has a tonal colouring effect, so only highly-specified, high-purity, listening- testing materials are suitable.
For example, in the Ankoru we use only Black Gate and Cerafine electrolytic capacitors for the audio circuitry. These caps eliminate the electrolytic mushiness without going over to the brashness of certain plastic caps. The valve selection was guided by the notion that the different sonic signatures of each type should be complimentary, leading to the goal of a sound that possesses both strength and finesse.
Before I go on to describe the Ankoru circuit in detail, I would like to say a few words about transformers and transformer coupling, since transformers play an important role in the design.
In any valve, waveform distortion is caused by the characteristic parameters of the valve changing in sympathy with the applied signal. In a standard RC coupled triode circuit, the valve is set up with a quiescent current (Iq) following through it and the load resistor, yielding a particular quiescent voltage on its anode (Vq).
With a negative-going input signal, the current is reduced and the anode swings positive due to the reduced voltage drop across the load resistor (R1 X Iq). The reverse is true with a positive-going input signal, the valve’s anode current is increased so the voltage on it reduces due to increased drop across R1.
There is a problem with this, however, because as the anode swings positive and the current decreases, the trans- conductance of a valve goes down due to the curvature of its characteristic. Of course, the reverse is true with a positive-going input signal, the transconductance goes up with the current.
This means that the positive part of the anode swing is compressed and the negative part is expanded – waveform distortion. Usually, this distortion only becomes serious with very non-linear valves and/or large voltage swings. When we want to drive a fairly meaty output valve, we need to swing a lot of volts because the mu of these types is necessarily low to keep loudspeaker damping up. In this circumstance, waveform distortion can easily rear its rather ugly head.
We need a system for keeping the current through the valve as constant as possible over the anode swing, i.e. a high load impedance. Increasing the load resistor on an RC coupled stage can only go so far, however. One soon runs into problems of resistor dissipation and PSU voltage if the anode current is kept at the optimum level.
The SRPP stage and his other active loaded cousins, such as the mu follower, have never really delivered the goods for me. Close listening reveals a lack of focus and immediacy compared to even the humble RC coupled stage. Anyway, SRPP is a feedback device and quite often that scheme doesn’t work very well electrically either, especially with the low impedance valves we would like to use as drivers. Simply pretending that you’ve got a low output impedance just doesn’t cut any ice in the world of real audio.
For large power valves, a low AC drive impedance is necessary because large valves have large and therefore highly capacitive grids. Thankfully, the low gain keeps down the Miller Effect, but it’s still there, so for good HF response, there is no getting around using a good low-impedance driver.
From the standpoint of sound quality, for a strong sound we need a beefy, low impedance driver. Wimpy driver equals wimpy sound. Drive two 845s with an ECC83 and it’ll be like putting a model aircraft engine in a Chevy Impala. Not exactly awe-inspiring.
The DC resistance of the grid circuit must also be kept low to control the effect of another rather annoying bugbear, grid current. Unfortunately, the vacuum in many modern valves is far from perfect, so there are quite a few gas ions floating around inside the bottle. Some of these ions will collide with the grid and draw electrons from the grid circuit. If the grid resistance is high, the grid bias will be modulated in tune with the signal, a real no-no in my book.
Also the grid may occasionally be driven positive on signal peaks, causing the grid cathode diode to conduct, rectifying the input voltage in the manner of a shunt diode supply with the decoupling cap as the reservoir. This action makes the bias voltage more negative, reducing the quiescent current through the tube, sometimes to the point where it will only conduct on peaks (Class C). In fact, a severe peak can cause the amp to cut off altogether, resulting in a total loss of output.
Worse still, the grid resistor/coupling cap combination as an RC time constant, so the effect lasts for some time after the overload has passed in sort of a time-delay distortion mechanism.
Reducing the grid resistor to combat these effects is no solution. We want a DC grid resistance similar in magnitude to the impedance of the driver valve, i.e. a few hundred ohms, not a few hundred kilohms.
Making your grid resistor 600 ohms will likely kill the driver stage and, anyway, would require a coupling capacitor so big that the RC time constant would put us right back where we started.
To cure the voltage swing problem requires a circuit element which has low DC drop but a high AC impedance. Plus, we need a low DC resistance in the grid of the output valve. And the device should efficiently couple the driver valve to the output tube’s grid.
The driver transformer is exactly what we need for the job. Its primary inductance presents an extremely high AC impedance to the driver valve and reflects the anode impedance of the driver into the grid circuit of the output valve. A good driver trans will have a primary and secondary DC resistance on the order of 300 ohms, so the problems associated with grid current are more or less eliminated. This is a resistance 1000 times lower than I’ve seen in some designs.
Ideally, the transformer secondary is left unloaded, i.e. there is no “damping resistor” put across it to cut ringing. An unloaded transformer sounds better and it gives the driver valve a higher impedance load.
There are two large-scale problems with driver transformers; HF frequency response and LF frequency response. These two requirements are mutually exclusive to a certain degree and many commercially available transformers sacrifice one for the other. The Tango transformers, for example, seem to go for impressive-looking HF specs but they have diminutive primary inductances which limit the LF performance.
The problem is compounded by the unbalanced DC current imposed by SE operation, which requires that the number of primary turns must be increased to counterbalance the loss of permeability caused by the air gap in the core. Leakage inductance is proportional to the square of the primary turns so it’s a real pain in the butt.
The driver transformer in the Ankoru has to handle 45 mA and still have superb bass, so it took some heavy calculator work and a few trees worth of paper to get it all working! [The Ankoru interstage trans will be available as a DIY part-ed.]
I love the sound of large triodes like the 211 and 845. The 845 was used in this amp because if offers greater power in Class A1. The 211 is a more voltage sensitive valve than the 845, its mu is higher but then so is its internal impedance. It can’t swing a lot of current at the low voltage end of the anode swing without having the grid driven positive into Class A2. When pushing the grid above zero volts, it no longer reacts as a high-impedance terminal. It starts to draw appreciable current, corrupting the input signal in a most unattractive way unless the driver impedance is extremely low.
The grid-cathode diode impedance of a 211 is about 2k, so we would need something around 100 times lower or hideous distortion would result. The waveform distortion could be corrected using feedback but why build an amplifier that is intrinsically non-linear?
The 845 can sink a lot more juice where the 211 starts wheezing, but since the mu is so low, it requires a driver stage capable of considerable voltage swing. The 845s in the Ankoru are biased at -100 to -200 volts for an anode current of 75 mA at 1200V B+, they look into a load impedance of around 6k, and put out a formidable 70 Watts. The output transformer has to cope with 150 mA DC and hold its 6k impedance down at LF, requiring a high primary inductance. This takes a serious hunk of iron, but the Ankoru output is just such a beast and the bass is awesome, if I do say so myself.
To keep the drive signal to the output valves clean requires a driver valve of excellent linearity. One could use an indirectly heated valve such as the 6BX7, very linear, or the slightly less linear 6BL7, but low impedance, low mu directly heated valves are definitely the best choice.
Since this amp has to be built using valves which will be available for some time into the future, so that replacements can be made throughout its life, it was necessary to use modern versions of either the 2A3 or 300B. I originally experimented with the 2A3 as I wanted a measure of its clarity and immediacy, but these valves have a very nasty habit of making toilet related noises even in the output stages of amps, and using one as a driver was impossible. I even tried some NOS samples but many were only marginally better, only the best and therefore rare and expensive samples were quiet.
So the 300B was chosen, and it brought its characteristic warmth and musicality to the amp as well as a greater impact to the bass. The 300B is operated with 300V across it and an anode current of 45 mA so it will last for ages, no more current or voltage was necessary for driving the 845s to full output. The 300B’s output is in fact so large that the 845s will be freaking (and so will your wife and the neighbours) before it runs into trouble, which makes its jobs and the job of the input stage easier.
Various input configurations were tried, all using the E182CC/7044 valve for its powerful sound. The original and best sounding configuration gave the amp so much gains as to be impractical. Long speaker leads acted like antennae and transmitted the amp’s output into the input leads causing instability. Super high quality cables and careful system set-up would eliminate the problem but as this is a commercial amp, it has to be dealer-proof, so a simple, single-stage RC coupled affair was settled upon. The 7044 was always run at a high current to really bring out its flavour.
The Ankoru is interfaced to the preamp via a coupling transformer to allow balanced operation and to properly ground the grid of the red hot 7044. A switching system permits regular unbalanced input as well. The Ankoru is intended for use with the Audio Note M3 which has output transformers and balanced outputs.
Ideally transformer coupling between the input stage and 300B would have been used but even super quality transformers impart a signature upon the sound (ultra mega quality ones don’t however) so a special copper foil capacitor with paper/oil dielectric was used to couple from 7044 to 300B. This capacitor, like all caps, has a sonic character but it was used to avoid a build-up of one type of timbre caused by the cascaded transformer coupled stages.
The power supplies are fairly standard, and of course valve not silicon, remember microprocessor parts in the power supply equals computerized sound. If you want your record collection to sound like a bunch of cheap CDs, then use silicon rectifiers for the audio PSUs like all the other junk in the shops. In fact, I would use valves for the filament supplies if I could – Tungar rectifiers such as the Ediswan 68506 would work for those who dare [Cool! – ed.] or AC straight from the mains trannie, but then punters would whinge about hum. I could have built gargantuan supplies which would have caused the primordial fires of a nuclear power station to die but this amp had to fit into an (almost) domestically acceptable chassis.
So a sensible but effective approach was taken; capacitor rather than choke input filters were used to get maximum voltage efficiency and chokes were used to get ripple down. The capacitors in the PSUs are directly in the signal path so they need to be of excellent quality and here the Cerafine types come into their own. They have a smooth and refined sound. Energy storage was not taken to extremes but the main HT for the 845s holds 50j of energy (the caps on the input side of the filter are isolated from the audio circuit by the choke and therefore don’t count).
It is necessary to have a rigid supply. Smaller caps generally sound a bit sweeter in the mid and treble, but if you want a decent bass quality, the last thing you want happening is the PSU flapping about all over the joint. You don’t put a lawnmower carburetor on a Ferrari engine.
Going for oil drum sized caps doesn’t work either (Question: Can you think of a trannie amp with super solid bass and complete and utter crap everything else?). Super sized capacitor supplies can pump out heap big LF current transients but they take heap big time to recover as well, and the impedance of the giant electrolytics just skyrockets as the frequency rises.
Regular capacitor power supplies integrate the demands placed on them so a bigger supply reacts a smaller amount but everything takes longer. So the PSU for the 845s is suitably scaled for an excellent all-round performance, solid bass through to sweet and delicate treble. Things are made a bit easier because the energy storage of a capacitance is proportional to the square of the voltage on it and at 1200 V it doesn’t take a big capacitance to store a lot of grunt.
To minimize the effects of the 845s on the preceding stages, the 300B and 7044 have their own supply from a separate mains transformer. Both are run from the same rail so that the 7044 has a really juicy supply to suck from, and remember the 300B is running into an unloaded transformer so there is minimal supply draw variation due to constant current operation.
The 845 supply is rectified with two 5R4s in a voltage doubler configuration to ease the peak inverse voltage requirement, the output impedance and peak current go up but it is still satisfactorily within the valve’s limits. The driver stage HT is via a 5Y3 rectifier and the bias supply uses a 6X5. The main HT is delayed by the bias supply, the driver stage and bias power is applied when the amp is switched on. The 6X5 is indirectly heated rectifier and so takes a little while to come up.
When the bias voltage reaches a safe valve the big 845 power transformer is switched in by a relay. If the bias fails for any reason, the relay will drop out, cutting the power to the 845s.
All in all I am pleased with the end result, the Ankoru when partnered with a good preamp such as the M3 and a good turntable, gives a superb musical performance. It can resolve the smallest nuances and subtle timbres of classical music and deliver the visceral impact of techno, even with relatively inefficient speakers.
In short, this was the design brief: A single ended amp which would give that SE charm and musicality but which would also send the big solid state boys back to their silicon shrines to have a serious rethink.

لینک های نمایشگاه RMAF 2011

http://www.stereophile.com/

http://www.positive-feedback.com/Issue58/rmaf2011.htm

http://dagogo.com/View-Event.asp?hEvent=37

http://audiofederation.com/blog/categories/shows/rmaf-2011/

http://www.6moons.com/industryfeatures/rmaf2011/1.html

http://jlvalin.zenfolio.com/   (گالری عکس های جاناتان والین)  http://jlvalin.zenfolio.com/p980229970

http://www.tonepublications.com/review/heavy-metal/

http://www.astrasuite.com/astrablog/2011/10/21/schimmel-voxativ-ampeggio-among-best-sound-at-rmaf-2011/

نمایشگاه های دیگری هم همین اواخر بوده که لینک هاشو گذاشتم :

http://dagogo.com/View-Event.asp?hEvent=36

http://www.6moons.com/industryfeatures/hk2011/1.html

http://www.6moons.com/industryfeatures/taves/1.html

http://www.6moons.com/industryfeatures/suisse2011/1.html

اینجا باید در مورد های فای بنویسم و نه از احساسات ام نسبت به بارون اما من این چند روزی که به این شدت بارون میومد خیلی حال کردم گفتم بنویسم بارون باحال ترین پدیده طبیعت هست.

حالا آفرینش چه کار خدا بوده چه همینجوری سرهم شده باید اعتراف کنیم واقعا زیباست و صدای بارون برای من مثل یک قطعه موسیقی گوش نوازه شایدم فراتر از یک قطعه موسیقی …

از های فای من بیخبرم ، هم از اخبار داخل هم از اخبار خارج …

خوش باشید

دوستی چند وقت پیش از شهرستان با من تماس گرفتند و قرار شد آمپلی فایری که ایشون ساختند رو من صداش رو بشنوم و البته این سری که به تهران تشریف آوردند قسمت نشد ببینمشون اما احتمالا در آینده یه قراری با ایشون داشته باشم.

این جوان بسیار مودب که اسمشون عارف هست سایت http://setson.persianblog.ir/ رو مدت زیادیه که راه اندازی کردند اما من قبلا به اشتباه فکر میکردم این سایت مربوط به شخص ای میشه که کارش کپی مطالب دیگران و به شکلی مقابله با کسانی است که به نوشتن در سایت های دیگر مشغول هستند.

من از این دوست بسیار خوبمون همینجا بخاطر اشتباهم عذرخواهی میکنم، از این اشتباه واقعا متاسفم و امیدوارم منو ببخشند.

این دوستمون چند وقت پیش که به تهران تشریف آوردند آمپلی فایر لامپی مبتنی بر 211 شون رو تو فضای شرکت آوین آوا تست کردند که گزارشش رو تو سایتشون گذاشتند :

http://setson.persianblog.ir/post/183/

http://setson.persianblog.ir/post/181/

چیزی که حداقل در ظاهر مشخصه سلیقه ایشون در ساخت هست و من همیشه ذهنیتم این بود یک آمپلی فایر دست ساز باید شکل و قیافه اش رو فاکتور گرفت اما ایشون خیلی خوش سلیقه و خوب این آمپ رو ساختند که بی شباهت به برند های تجاری بازار نیست.

من به ایشون بابت زحمتی که کشیدند تبریک میگم و امیدوارم تو کارشون موفق باشند. اگر قسمت شد و صدای سیستمشون رو در آینده شنیدم حتما تحلیلی هم تو سایتم خواهم نوشت.