Koji 24bit DAC? (split iz SissySIT)

[Lemić]

9 hours ago, vladd said:

 

Pregledati koja sve napajanja ima taj dacić, možda porazmisliti o kompleksnijem konektoru za napajanja, pa izvesti svako napajanje posebno, odvojeno iz spoljne kutije, a interno rasporediti žičkama gde treba.

Ostaviti tantale na digitaliji, i pokoji MLCC, multilejer, tu elko nema uspeha ako nije smps ispred njega, mada iako low esr, veći su otpori nego kod spomenutih na tim frekvencijama potrošača od tantala ili polimera.

Sporom analognom delu odgovaraju spora linearna napajanja, dotične popularne Elne, mada bih ja osobno, laku prednost dao Nichiconima i Panasonic brendu. Onako na ukus bez dokaza  

Hvala, no, uhuhuh, mislim da ipak ne bih "terao" Gosn Nebojšu toliko duboko u digitaliju,  ako bi se neko od ovdašnjih prihvatio te rabote, što da ne, podrazumeva se uz nadoknadu, hoće li ko?

Ima kod Toce Elne Simlic 2 kao i Nichicona UFG Gold i UFW, za konačni predlog bi onda bilo potrebno rastumačiti šemu, ja nisam taj, znači da i za to mora neko, čevapi kao materijalni dodatak.

[NIXIE]

@Lemić, možda bi PCM63 kod tebe zvučao previše agresivno, nije isključeno da se taj E30 malo dorađen ili AD1865 recimo pokažu kao humanija komponenta za slušanje sa Lauterima. Ono što već znamo je da AD1865 ima više basa od svih koje smo do sada slušali, pravljenih i fabričkih, i to dobro definisanog. PCM63 je čistiji u ostatku spektra. AD1865 se naročito dobro uklapa sa FR zvučnicima iz poznatih razloga. Zato ja još nisam načisto koji će trajno ostati dok ne napravim oba i probam ih u istom okruženju, prvenstveno mislim na napajanje. Znači sve će biti isto, samo menjam DAC pločice. Tako je test mnogo korektniji. Ovaj koji smo mi pravili nije skroz fer bio prema AD1865 jer je isti imao malo skromnije (i dosta jeftinije) napajanje od PCM63.

Izmenjeno Novembar 12, 2022 od NIXIE

[Lemić]

Ovako, što kraće mogu. Lowther jeste problematičan zvučnik u pogledu agresivnosti, čuveni "Lowther shoot", nisam imao prilike da slušam razne kabinete, ali je izgleda TQWT princip u Teresonic izvedbi najbolji izbor, opire se gotovo svakom pojačivaču, bilo tranzistorcu, bilo lampašu, tek sa SE GM 70 sam ga ukrotio. Zatim, modifikovan je drajver, Branko Bin je uradio završnu i najznačjniju korekciju, ostao je isti frekventni doseg gore, ali ispeglana agresivnost i isplivao bas, sada se rade pod varijacije, imam još jedan par drajvera koji izbacuju više basa nego što mi prija, znam da to suludo zvuči, da Lowther ima višak basa, ali za moj ukus je tako i drajveri su na polici. Doduše, to je sa D10, još ih nisam slušao sa E30II, uskoro ću morati jer je treći par spreman za preradu i moram da odlučim u kom smeru da se krene, a inače će biti od 15 Oma, što će u mnogome prijati izlaznom trafou pojačivača, iako trenutno bas može da zatrese ambijent, što je inače neuobičajeno za bilo koji FR zvučnik, naročito za Lowther, ali ovo kod mene i nije više tipičan.

Sama želja za prepravkom E30II, nije proistekla, ili vođenja zbog nekog nedostatka, već zbog moje nemirnosti, da nikad ništa nije toliko dobro a da ne  može bolje. Ko što sam već naveo, osim dva elkosa od 1.000mf, svi ostali nisu ništa više od korektnog, jedan deluje da se već naduo, Gosn Buca prvi komšija i uvek voljan da majstora, iako punih 80, klinci bi mu pozavideli na agilnosti, umešnosti i istrajnosti, ne staje dok ne završi započeto. Znači, kockice su se skockale, barem za elementarno poboljšanje prostom zamenom elkosa, koji su verovatno najlošiji nužni elemenat u ovoj našoj audiomaniji. 

[Mikorist]

pre 2 časa, NIXIE reče

@Lemić, možda bi PCM63 kod tebe zvučao previše agresivno, nije isključeno da se taj E30 malo dorađen ili AD1865 recimo pokažu kao humanija komponenta za slušanje sa Lauterima. Ono što već znamo je da AD1865 ima više basa od svih koje smo do sada slušali, pravljenih i fabričkih, i to dobro definisanog. PCM63 je čistiji u ostatku spektra. AD1865 se naročito dobro uklapa sa FR zvučnicima iz poznatih razloga. Zato ja još nisam načisto koji će trajno ostati dok ne napravim oba i probam ih u istom okruženju, prvenstveno mislim na napajanje. Znači sve će biti isto, samo menjam DAC pločice. Tako je test mnogo korektniji. Ovaj koji smo mi pravili nije skroz fer bio prema AD1865 jer je isti imao malo skromnije (i dosta jeftinije) napajanje od PCM63.

Treba da sad čuješ BASS sa PCM63P-Y . Ovo nije normalno ., Stvar je usviravanja.

[Mikorist]

pre 5 časa, shonne reče

Ne idi, ne idi, ne idi, neeeee 😁😛

Poenta je u tome sto 70% ljudi ovde na forumu hoce od sprave koja nominalno kosta 100 ojra da dobiju iz neke budzevine najmanje 2k ojra, 
kao i uvek tvrde pazar, glume ludilo i prave se najpametniji. 

Paaaaaa stara ona narodna - nema pite od govana. Jbga na recima, ali je tako.

I na kraju sam ja kriv za to i niko mi ne veruje. Asve sam gologuzan. Krpe se gaće da se ne vide dlake na dupetu.

 Msm bambara mi nesto.

moze i ezoterija, ako znas sta radis i sam radis i imas lovu

ako es na oruk, moze i tako, ako si zasluzio

al oni koji zasluzuju, takvi ne smaraju na forumima,  a ovi drugi  nikad nisu zadovoljni odgovorom

[shonne]

Miko, samo opusteno. Sve je ovo zabava 🤗

[Lemić]

Mirkane, ja sam prvi CD plejer dobio na poklon, neki Tehniks od 1.500 Dojče maraka i to više godina od pojave digitalije. Zatim sam kupio jedan od uopšte prvih parova plejera i konvertera, Audio Alhemi, menjao operacione, još ih imam negde, pa drugi dak, pa slušo niz spravica i na kraju uzeh Philps 14 bita. Lep ton ali fali malo dinamike, pa uzeh polovan Kembriž sa 1541 za 1.000 maraka od jednog viđenijeg trgovca. Pošto smo se znali, preredio mi je malu sesiju, Unisoni i Sonus Faberi, prvo Kembridž pa onda Wadia od 6.500 maraka, bila i od 10 hiljada, taj model samo imo podešavanje glasnosti preko daljinskog. Ismejo sam se, Wadia lošije zvučala u odnosu na Kembridž. Međutim, nisam mogao nešto da ga varim i dao sam ga bukvalno posle mesec dana i vratio se na Philips. Zaključio sam da je digitalija jedna obična prevara i da je besmislica davati velike pare i nikad više nisam ni pomislio da bilo šta skupo kupim, iako sam imao prilike da čujem dosta dobrih i skupih sprava. Uzeo sam Philips 304 MK2 za 200maraka, iščačko ga i to je jedino bilo prihvatljivo za moj ukus, nisam hteo da uzmem ni radni primerak Rogićevog Audiala 1 po skromnoj ceni, do Toppinga, koji sam uzeo da bih koristio PC i smanjio rezanje. Slažem se da ne može pita od lajna, ali isto tako ima pita koje ko da su baš od lajna, samo za skupe pare. Da, u to neko vreme je izašao DA čip od 32 bita, AD, velik ko kutija cigara, objavljen je u Stereoplay intervju sa njegovim konstruktorom u kome je on izneo otprilike ovako, " nije problem da se napravi i sa više bitova, ali je sasvim dovoljno i 14 bita". To je u stvari meni stavilo tačku na digitaliju. Inače, imam vrh analogiju, što bi se reklo, sa jedno hiljadu probranih ploča, sve se više omišljam da otuđim, ali u digtalno ne bih uložio ni desetinu kolko košta komplet vrteška. Naravno, merak nema cenu i ništa protiv, naprotiv.

Izmenjeno Novembar 13, 2022 od Lemić

[vladd]

1 hour ago, Lemić said:

Zaključio sam da je digitalija jedna obična prevara i da je besmislica davati velike pare i nikad više nisam ni pomislio da bilo šta skupo kupim, iako sam imao prilike da čujem dosta dobrih i skupih sprava

Ne bih se baš tako grubo izrazio, ali je digitalija nastala i opstala zarad lakoće i masovnosti širenja muzičke industrije uz neki kompromisni kvalitet značajno bolji od recimo tjunera i u rangu komercijalnih sprava. Nije baš bila za ezoteriju, ali eto vešti trgovci uvežbaše pričanije.

Odsekoše sve preko 20kHz, Nikvist kažu, ali Nikvistov minimum(bože, jel opet odabraše minimum zarad profita) važi za ČIST sinus, a da, dosetiše se da je kompleksni signal splet sinusa raznijeh frekvencija, ali otsekoše suvišne harmonike na jeftino i komercijalno. Brzi konvertori su bili preskupi, a sa tim i profitno nepopularni. Prvi koji su bili odbačeni profesionalno su R2R, zarad gadnog šuma, ali šta fali za potrošnju audio konzumerizma, masovno šum i ne čuju...  

Iako su i trake i gramone bile poželjne, dobre i skupe, kada znaju da odsviraju i preko 30kHz.

Najbiserniji su entuzijasti koji na tu, eliptičnim oštrim filterom odsečenog opsega digitaliju, dokupiše opijeni pazarom i supertvitere.

Znalo se da ultrazvuk utiče na sakulu, ne ulazi direktno na kohlearni mehanizam već preko kostiju, sve u svemu ima uticaja na utisak...tako da otsečenih 20kHz su prc, pablik ripablik of jeftino...

Na to se dodade i najveseliji part, muzikalni protokol za USB, statistički ima toliko grešaka da ne utiče puno, samo malo, tzv podnošljivo prihvatljivo, ko će to da čuje filozofija, malo mnogo više uticaja nego materijal konda, kabla, vrste tranzistora i slicne perfekcijice, a zašto je tako, pa je nevredno pomena, zato što je jeftino i komercijalno za maskonzumaciju.

I slažem se potpuno da spravac od 100 evra ne može da svira kao pomagalo od 2000, ali, može spravac da se namontira da nije 10-20 puta inferiorniji, nego pomalo, za nijansu.

Isporuka novca profitu nije cilj diy, već fokusirano uvećanje objektivne upotrebne vrednosti sprave tehničkim poboljšanjima, ne bajkama.

Iako kod muzike bajke imaju nekakvu prođu opšteg utiska, kod neke konkretnije inženjerije su besmislene, auto koji ide 200 ide 200, nema tu "osecam da ide 350", ističe mu se čini mi se....  

 

 

Izmenjeno Novembar 13, 2022 od vladd

[Lemić]

Ispadoh grub, iako nisam želeo ni mislio tako, jer ne bih uostalom slušo digitaliju i sad zamlaćivao forum prepravkom dakića. Dobro si napisao, nijansu slabije jeftini konverter u odnosu na mnogostruko skuplji. Sećam se tog prvog daka, DAC in the box, nađem ideju o prepravci, sklop od nekih 160 tranzistora, budalaština. Problem konvertera proističe zbog sirmaštva i praktično mora da odsvira nešto nepostojeće po nekom svom algoritmu, to sam davno pročitao i jeste tako, zvuči prazno i agresivno i to je ono što smeta. Međutim, Topping, posebno ovaj novi, nema tu anomaliju, vešto je prepokrivena iako i dalje spada u donju cenovnu klasu. Ipak digitalija dugo traje već, ne samo u audio sferi već u čitavoj elektronici i sasvim je normalno da postojo progres koji se eto, primenjuje i u obradi zvuka. Kad se pogledaju reizdanja i nova, retko koji vinil da nije prošao neki vid digitalne obrade. Komfor je bio presudan, imam giga i giga flakova i to je neprocenjivo i zato se mora malo i zažmureti i na oko i na uvo. Kad vratim film, ovaj DAC, koji je nekoliko puta jeftiniji od prvog mog Alhemija, svira opet više puta bolje, tako da je razlika u pozitivi neuporediva. Onda je Meridian važio za melozvučan konverter, košto muški, sada ga ovi Toppinzi nadmašuju samo tako, a čujem da i masa tih Kineščića lepo zvuči i vrlo su popularni. Naravno, ovi testeri drže monopol i neće da ih uzimaju u obzir. Ne znam dal znate, uobičajeno je dati na poklon testirani uređaj, iako zvanično ne postoji tarifa, recezent to posle proda i tako naplaćuje svoj medijski uticaj koji proizvođači komercijalizuju, sve je igra oko novca, što je sve skuplje, više ga je i samim tim i profit. Prosto zar ne.

Izmenjeno Novembar 13, 2022 od Lemić

[Mikorist]

pre 39 minuta, vladd reče

Odsekoše sve preko 20kHz, Nikvist kažu, ali Nikvistov minimum(bože, jel opet odabraše minimum zarad profita) važi za ČIST sinus, a da, dosetiše se da je kompleksni signal splet sinusa raznijeh frekvencija, ali otsekoše suvišne harmonike na jeftino i komercijalno. Brzi konvertori su bili preskupi, a sa tim i profitno nepopularni. Prvi koji su bili odbačeni profesionalno su R2R, zarad gadnog šuma, ali šta fali za potrošnju audio konzumerizma, masovno šum i ne čuju...  

 

Kod Mirovog DAC i Pavouk ne postoji Nikvistov minimum. Nema filtera.... Zapravo ja sam stavio da mi gađa do 9Mhz kroz konverter. Mož da izbaciš skroz kond paralelni ali onda mož da se desi da imaš oscilacije i da hvataš prvi program radio Beograda kroz DAC 

[Mikorist]

Još se bolje čuje kad ta ultrasonična energija prođe kroz Iron - negde na trafou se odseca - i neko čudo se događa sa prostorom - širi se umesto da se sakuplja kao kod DSD

[vladd]

Po Nikvistu je digitalizovan analogni materijal i tako digitalizovan gurnut na ulaz u DAC.

Iza daca je "željeni" signal u prvoj Nikvistovoj zoni, a "duhovi" u drugoj i trećoj, i nastaje čuvena sinx/x distorzija, ako se ostavi klot i ne modeluje.

Figure 3. This representation of a DAC output in the frequency domain shows that the desired signal is generally within the first Nyquist zone, but many image signals are present at higher frequencies.

The sin(x)/x (sinc) function is well known in digital signal processing. For DACs, the input is an impulse and the output is a constant-voltage pulse with update period of 1/f_S (the impulse response), whose amplitude changes abruptly in response to the next impulse at the input.

Izmenjeno Novembar 13, 2022 od vladd

[Mikorist]

Non-oversampling Digital filter-less DAC Concept by Ryohei Kusunoki Nov. 1996   To Confirm the Original 44.1kHz/16bit Format      It is exciting to create something new. Some people just grab a soldering iron, others deliberately start a simulation. Everyone seems to have his/her own approach. In my case, it starts with going back to the basics, research the history, and re-construct the whole picture in my mind. On starting this project I did research as many resources as I could put my hands on.      As the new generation of CD format is appearing on the horizon, I thought the basic concept should be "To Confirm the Original 44.1kHz/16bit Format". A CD in our hands has exactly the same data, every bit to bit, as to the one that left the studio. To recall this dreamy fact, the above theme would be quite appropriate. Any high-bit or high-sampling does not have its raison d'être unless it surpasses this level of accuracy. About Non-Oversampling      After examining the following two aspects, I came to a conclusion that 'it is quite difficult to carry out oversampling as theoretically under the current technology'. 1) Oversampling and Jitter      There are two axes on digitizing the sound. The time axis and the amplitude axis. In case of CD, they are 44.1kHz and 16bit. In other words, we have to press in the amplitude data into one of the 16bit stage at every 22.7 s. That produces maximum of +0.5 LSB error, and the digital audio starts by accepting this error at the beginning. However, this error only concerns the amplitude axis and no amount of error was admitted on the time axis. Let me suppose that the accuracy of 16bit means how accurately the acoustic energy (time x amplitude) is transmitted by being distributed into each steps of 16bit. Then, by making the amplitude data more accurate, we can distribute the error onto the time axis.      If we distribute 1/2 of the error, 1 ÷ 44.1kHz ÷ 216; ÷ 2 = 173 (ps)      This represents the maximum limit of the acceptable error (maximum limit of the jitter). (diagram 1) [diagram1] acceptable error of 44.1/16bit [diagram2] acceptable error of 8 x sampling/20bit      All of the above is based on the basic sampling rate. When in 8 x oversampling and 20bit, that number would be 1.35ps (diagram 2). This is a totally impossible number to achieve for a separate type DAC which has to recover the clock by PLL. This means that under an average jitter environment, the oversampling can not operate theoretically, and lowers the accuracy within the operating field. In short, just by oversampling the original data, 16bit accuracy can not be satisfied anymore. 2) Oversampling and High-Bit      Originally, oversampling was developed to allow the use of an analog filter with gentler characteristics as a post-filter, and not to increase the amount of information. Many people still misunderstand this. [diagram3] principle of FIR type digital filter      The principle of the most popular FIR type digital filter is to shift the original data and overlay them together, not to create an additional one (diagram 3).When it overlays the data by multiplying the coefficient to the original data, there appears new information below 16bit and to recover this finer information, we need a higher bit rate processing. For example, in case of a high-performance digital filter SM5842, this processing is done in 32bit and the filter round them up to 20bit to the output, creating more errors in the re-quantizing process. Recently, this problem was dealt with and a filter was created which can produce 8 x sampling all at once. But even with that, as long as you can't output the internal word length as it is, there's no way you can prevent the errors to occur.      It may sound contrary, but if you take this error into account, 16bit without oversampling is more accurate than 8x-oversampling/20bit. [diagram4] image noise continuation      Then what is going to happen if you eliminate the oversampling process? Theoretically, the image noise will be repeated infinitely to higher frequencies (diagram 4), and a conventional answer would be 'it will sound awful'. Really? This has nothing to do with the "Shannon's theorem", nor do I intend to challenge that. Shannon's theorem considers a sampling theory on transmitting an information. I am talking about the perception of the information. That is, if I must say, "the limitation of our auditory sense is a powerful low-pass-filter and the Shannon's theorem is satisfied at the echelon of human auditory perception." My challenge is rather toward those who listen to the sound through theories and oscilloscopes.      Another way of thinking is that, even if humans can't hear it, the equipment that follows can and will be affected by it.      However, 8x-oversampling/digital-filter can only cut off the frequencies between 22.05kHz and 330kHz. Everything beyond 330kHz is all coming through untouched, meaning the degree of effect is determined by how the said equipment reacts to the ingredients beyond 330kHz. My guess is, if 100kHz signwave comes through, there won't be any problem. Problems of the Digital Filter      The diagram 5 shows the principle of the most popular FIR type digital-filter. The "T" represents a delay circuit for each sampling interval, "a" is for the coefficient multiplier, and "+" is an adder. After delaying the input data, it multiplies with the coefficient, and this process is repeated n times. This 'n' is called the number of taps. The more taps it has, the higher the performance of the filter is supposed to be. The delay mentioned above is not that of a calculating time, but more like a waiting time until the next data arrives. T:adelay circuit for each sampling interval a:coefficient multiplier +:adder [diagram5] FIR type digital filter [diagram6] FIR type digital filter (in case of SM5842)      It is rather hard to understand this diagram instinctively. It didn't hit home with me, either. But, one day, it occur to me to replace it with the equivalent of the reproducing hardware system. (diagram 6). The delay circuit is replaced with that of the delay of speed of sound, the multipliers with the attenuators, and the adding is synthesized in the space. The number of the speakers corresponds to that of taps. The diagram shows, as an example, the computation of CD data through the high-performance digital-filter SM5842. The accompanied numbers are the actual sizes in the space when replaced with the hard-ware. Since the sampling frequency of CD is 44.1kHz, each delay time for the 1 x sampling is 22.ms per tap. To achieve 8 x sampling, SM5842 repeats 2 x sampling three times, and each step incorporates the taps of, 169 degrees for 2 x, 29 degrees for 4 x, and 17 degrees for 8 x. The accumulated delay of each step becomes, 1.92ms, 0.16ms, and 0.05ms: total of 2.13ms.      Our auditory sense does the frequency analysis at every 2ms interval, and 2.13ms of delay can be caught by our ear.      If the speed of sound is 346m/s, the total length of the row of speakers becomes 737mm. ( In the diagram, the distance between each speaker is presented by the total delay divided by the total number of taps.)      Now, you can imagine what kind of sound will result from such a system. All the notes coming from the speakers before and behind, will mix, intervene with each other, and spread. I would like to express this expansion of the sound over the time axis as a "diffusion of sound coherence". For example, if an attack of a piano note was not clear enough, as if the felt on the hammer became thicker, you might be hearing this "diffusion of sound coherence".      We also need to consider this issue not only on the playback systems, but more totally, including the recording systems. [daiagram7] in case SM5815A is used in 1/2 decimation The diagram 7 indicates the diaphragm 5 replaced with a recording hardware. If you ever felt the digital recording somewhat lacking a core of the sound, please examine this illustration carefully. In a way, one point recording using digital filter is so much nonsense. The time will come in the near future when the performance of a digital filter will be evaluated not only by its cut-off characteristics but also how small a number of taps it has. If the digital filter is a necessary evil, we have to make sure to limit the total delay within 2ms throughout the recording and playback so that it won't be caught by human auditory sense. The Sound of Non-oversampling      We can control the "diffusion of sound coherence" only by constructing it with smaller number of taps. From that aspect, Wadia's decoding computer (13 taps) or Luxman's former fluency DAC, DA-07 (3 taps) are considered to be excellent machines. They both received (Wadia still does) outstanding appraisals at the time for their sensual representation of the sound. The sound of non-oversampling DAC is on the extension of these machines, and theoretically, it can exceed those achievements.      The difference between the non-oversampling DAC and the conventional DAC with the digital filter lies whether you attach importance on the accuracy in the time domain or in the frequency domain. In other words, whether you choose the musical performance or the quality of a sound. This trade-off line defines the boundary of the current digital audio format .      A natural, stress-free sound that communicates the musicians' intention directly to you. That is the sound of non-oversampling DAC. The feel of this sound is closer to that of analog reproduction. Introducing Non-PLL clock      We can still hear the characteristics of each different transport even after lowering the jitter sensitivity to a minimum by non-oversampling. This is an incomplete section of today's digital audio format. The fundamental advantage of being digital, that its quality does not depend on the conveying form, falls completely short here. That is because we have to create the time axis from the incoming data by PLL at the receiving end. This is a flow inevitable to the current DAI format which requires word-synchronization. It is often misunderstood as if the time axis is digital, instead of analog, because it receives discrete value after sampling, but actually, it is completely analog. When the time axis is distorted, the analog wave form is distorted with it.      Then what is going to happen if we read it by its own clock at the receiving side? Unfortunately, the word-synch can not be kept and the waveform will be broken into shreds. I tried to re-clock it with a separate non-PLL-clock after once it's locked with PLL and reproduced (diagram 8). By doing this, the fluctuations within 1 clock of the PLL are completely absorbed and not transferred to the reproduced waveform. But any fluctuations beyond 1 clock, even if it'[s only 1ps, are magnified into that of 1 clock. This happens often, because of the frequency difference between the PLL clock and non-PLL clock. In case of this 50MHz non-PLL clock, the amount is 20ns per every 0.1ms. This is, after all, more than 100 times of the 16bit criterion. 1. Non-PLL clocl 50MHz 2. PLL clock 2.8224MHz(44.1kHz x 64fs) 3. re-clock pulse [daiagram8] Non-PLL re-clock      How does it sound? Are the notes broken or jumping around and unbearable to listen to? Somehow it does not, and not only that; it generates an extremely realistic sound field. A certain acoustic atmosphere envelops the room making you feel like you are on the same floor with the performers, communicating the tensions and relaxation among musicians to you.      This experience made me wonder if human ears are somewhat insensible to jitter. Whether it has a large or a small amount of jitter is not really an issue. The real problem is the constant fluctuation of the time axis caused by PLL. What is more important is the structure, including the time axis, of jitter, rather than the amount of it.      However, we can still hear the different characteristics of transports. I suspect that we may be hearing an effect of the original jitter detected by the non-PLL clock and entwined around the beat component. Measurements      The diagram 9 indicates the frequency characteristics (with the emphasis on and off). It looks a tube amplifier measurements. The roll off at lower frequencies seems caused by narrow frequency range of the analyzer itself, because I get the same response with whatever CD player I'm measuring (the manufacturer's spec sheet claims 20Hz~100kHz). The fall at the higher frequencies are caused by the aperture effect.

[daiagram9] Frequency

[daiagram11] Conventional DAC

The diagram 10 indicates 1kHz sign wave, -20db. For a comparison, the same waveform through a conventional DAC with digital-filter is shown on the diagram 11. The obvious notches that remained are all formed by components beyond 20kHz and could not be detected by human ear.

 

[daiagram12] Non-oversampling DAC

[daiagram13] Conventional DAC

20kHz, 0db is indicated in diagram 12, and that of the comparing DAC in diagram 13. In diagram12, it seems like 22kHz square wave is under amplitude modulation at 4kHz, and no 20kHz can be seen. I am not sure if this is perceived as 20kHz when it is filtered through our auditory sense. I would like to hear opinions from psycho-acoustic professionals. Considering the limit of impairment perception of humans (around 200Hz), however, this amplitude modulation of 4kHz does not need to be worried about at all.

Non-oversampling DAC Conventional DAC

[daiagram14] Ful-bit inpulse

The inpulse response is shown at the diagram 14. The comparing DAC's response is juxtaposed in the same diagram. The one with the inpulse shown downward is that of the non-oversampling DAC. With passive I/V conversion, unless you invert the data somewhere along the way, it comes out as opposite phase. While the comparing DAC shows a familiar waveform, non-oversampling DAC indicates an excellent pulse response. The slant of the top (bottom?) is caused by a low pass filter (160kHz at the time of the measurement). The pre-post echo shown at the bottom picture indicates the "diffusion of sound coherence". I am not saying that you hear this echo as it is, but suggesting that the process which produce this waveform has a problem in itself. If you examine this waveform more closely, undulations of longer term should be observed ahead and behind each echo.      * The diagrams here are of Mr. Kusunoki's proto-type DAC and not of the PROGRRESION converter. A Comment on the New Formats      The relationship between the sound and the measurement still remain mysterious. You cannot achieve good sound just by competing on the numbers of zero over the distortion factor nor by excessively extending the frequency range. However, in the next generation digital format offered today, the selling points for better sound are quantizing bit numbers and sampling frequency rates. It only means lowering of distortions and extension of frequency range.      The appearance of CD was an epoch-making event as a new format to follow LP. It delivered the sound of the master tape to our listening room. It was a crystallization of efforts of the engineers of that time. Compared to that, the new generation CDs offered today only concern raising the data rate; something similar to the idea of EL cassettes. The life span of the format will be a very short one. What we need is to accurately understand the merits and demerits of the current format and create the new one that pertinently matches our auditory sense.   This is an edited version of Mr. Kusunoki's three part article published in MJ magazine from Nov. 1996 through Dec. 1997. transration by Yoshi and Irene Segoshi

[Mikorist]

Mogu da potpišem ovo što je napisao gore japanac da čuje da je tačno. Neko čudo se događa sa prostorom i kašnjenjima da peva kao na stereoidima.

Hakovan DAC čip prevedeno. Nateran da radi izvan svake logike i manuala.

 

[vladd]

Japanac ima izlete u lupetanja najcrnje audiofilske madžikorame. Jeste, Šanon je za informaciju a zvuk je šta, magija...a zapravo digitalizacija zvuka opstaje u low cost domenu, za razliku od pipeline ADDA arhitekture koja se koristi u skupoj mernoj opremi...delta sigma je za multimetre da se ne pomisli...

BTW on filozofira na zahvatu oversemplinga kao da će to poboljšati i samu AD konverziju koja je odrađena po Šanon- Nikvistu. Tu mema popravke, materijal je digitalizovan i takav memorisan.

Takođe uzgred, on je to iskombinovao 95-e, pa par decenija to nikoga nije zanimalo. Ali manje bitno. Ogavno je što aludira na (samo)proklamovane audio feudalce, koji su kao odabrani da "čuju" neštoštagodizmaštali, a nisu ni dirigenti ni zvezde muzike sa tim talentom, pa ko ne čuje i ne prizna taj je magarac, za ostale konzumente kao nije bitno(njima dosta sprava od 100$)...previše plitkih manipulacija i zavođenja zanesenjaka. Word dropping tehničkih detalja a onda polivanje tehnike mokraćom vlastele pune emocija...važilo bi da je dobio bar IEEE nagradu, ne mora Nobel, ali nije, izgleda samo ZEWA plaketu...

Neka oversampluje, ali neka ne spominje vinil i traku, nema tu AD signal vako kako je rađen u studiju šta da traži

Elem, oversampling postojeće digitalije je dobra stvar iz nekoliko razloga, nebitno za analizu, većinu ne zanima, čita se samo milozvučno predrasudama, ali je i to deo uživancije pa neka i treba.

 

Izmenjeno Novembar 13, 2022 od vladd

[Mikorist]

pre 26 minuta, Mikorist reče

Hakovan DAC čip prevedeno. Nateran da radi izvan svake logike i manuala.

I tu opet možemo da se svađamo danima da ne valja kockast signal. Da ne valjaju merenja. Da šušti.

Da Vladd i Majda prave opservacije do Nove Godine 

[Mikorist]

pre 2 minuta, vladd reče

Japanac ima izlete

Da nije bilo japanca ne bi Pavouk napravio DAC - ni DDDAC ne bi postojao. Svi ga citiraju da je njegova zamisao.

A onda ni Miro njegovu verziju.... Ovo je sve njegova ideja.

[NIXIE]

Subjektivno je potvrđeno da mnogi DAC čipovi bolje zvuče u NOS modu, mislim oni koji se mogu naterati tako da rade. Šta tehnički iza toga stoji, jeste bitno, ali i tu kao što vidimo ima raznih mišljenja kako se to odražava na sam zvuk, opet kažem čisto subjektivno. Teško je naterati bilo koga da sluša tehnički i merno superioran uređaj, ako nađe neki manje tehnički savršen koji mu bolje zvuči.

[vladd]

Nema tu šta da se meri, šteta paliti opremu.

Poenta je da poneki i relativno jeftini dac, sa diy unapređenjima proizvodnih zakidanja, može da završi posao, a ko voli nešto treće, široko je polje.

48 minutes ago, NIXIE said:

Subjektivno je potvrđeno da mnogi DAC čipovi bolje zvuče u NOS modu

Potpuno legitimno, i pitanje je šta ta neka clock- jitter- noise izbijanja u ultrazvuku mogu da urade sa utiskom slušanja, zapravo u toj zoni slušni aparat ne razlikuje informaciju, ali je nerv "predgrejan", reklo bi se, dodato mu je prednaprezanje, ofset

Svakako da je to zona za igranje, nema nikakve obaveze da se traži perfekcija signala, već da se istražuje perfekcija utiska, sa jasnim usmerenjem

Nije šum i distorzija zona outlaw, već deluje, bar meni, čudan trud da se proglašava jedan šum za nevaljao a drugi da je plemenit, više rase...šumizam neki 

U tom slučaju, ja sam za vinil rasu, uber alles

[Brijac]

Vinil mi je na listi, za nekih 20 godina kada je u planu penzija, do tad postoje bitnije stvari za ulaganje 😁 Skuplja dara nego mera u ovom trenutku zivota. 

[vladd]

E pa prijatelju, meni penzija za 5 dana(slovima pet).

Zna se za koji tim treba-mora da navijam

[Mikorist]

Pukli smo prevedeno.