ralph

I was recently made aware of a customer to this thread. After looking across the posts, a few comments: In my opinion this apprehension lacks any real basis. Afi+USB does not use optocouplers. It uses professional fiber optic transmitters. When operated within specifications regarding temperature and current – and that's the case – the LEDs in the transmitters will work as long as LEDs normally work and that is almost eternal. In my whole life I have never seen a broken LED. I go so far as to give a lifetime replacement warranty for the fiber optic transmitters. So sure am I that this will never happen. Moreover, even if the transmitters are very old, the LEDs do not just broken. You only lose intensity. Since the transmitters are however designed for 70 meters and we use only 2 or 6 meters, the margin is huge, before no signal arrives. In addition, the signal quality on the fiber optic line is completely irrelevant to the sound quality, if only just good enough to properly differentiate between 0 and 1. The really exceptional of the Afi+USB design is I think not yet clear. Here no USB is cleaned. Afi+USB works as a real source device, together with a computer that delivers the audio data. Inside of the Afi there are two reference oscillators which are clocking the audio data stream under ideal conditions and, because of the fiber optical connection to the USB interface, isolated from any influence by the computer as good as possible. The small external USB interface quasi only provides the data to the Afi. All things important for sound are done in Afi. As far as I can see, Sonore makes in this respect nothing else like many others. This basically seems to be the same concept as in the MC3+USB. In my view the main drawback is there that the USB receiver is sitting on the same board in the same case. Also in my experience the usual electronic components for galvanic isolation do not shield RF interference good enough. RF has the nasty habit, not to stick to physically existing conductors. Therefore I felt the design concept of Afi+USB – two spatially separated devices and fiber optics – is important for very clean islolation of all computer technology from the audio processing. More info on our website: afi(s) - USB module But I am sorry, not all sites are available in English language at the moment and photographs will only be added during next week. Requests by email to me ([email protected]) are very welcome. Best wishes Ralf ACOUSENCE

Hi Fred, I trust you this experiences. But do you really know what you have compared? Have you made the recordings by your own? Do you really have material of the same recording in DSD and PCM, and are you absolutely sure that not one of those is a result of a conversion process outgoing from the other. With prerecorded material it is difficult to separate the influence of the recording format from all other influences. Therefore I check converters always with a repeatable high quality analog signal. Then I take a AD and DA conversion line. Now for me the perfect ADDA converters are converters I can not hear. This means no difference if I switch between analog direct and ADDA output. If I do this test with one of the highly recommended DSD converters I hear much difference. As much difference as I never heard before since early CD days. But the difference is completely different from the influence of a standard sampling PCM line. The signal seems to be polished in a special way. I think this is the explanation why many people like SACD. It has to do with listening habbits and expectations. And I think all people that come from a very analytical and technical point of few like SACD very much. For me the artistic side of the recording process is the most important thing. I want to reproduce the intention of the performers, the atmosphere and the musical liveliness in a best possible way. Regarding this the output of the DSD line was very disastrous. This test did also show that DSD with 128fs is quite close to the ideal. But 192kHz was it a bit more, and this with converters for about a quater of the price of the DSD converters. One remark: I must admit that this test is based on a test with only two devices. I don´t want to specify the devices here, but I used the DSD converters that many DSD recording people highly recommend as the best. I hear the same characteristic on nearly all SACDs and so I´m convinced that this is a systemic characteristic. On the other side it is certainly no problem to find a worse sounding PCM converter. I searched a long time till I have found the one we use for all recordings for a few years. Best wishes, Ralf P.S.: But please let us stop this discussion, it makes no real sense. DSD and SACD is dead. Perhaps there are some labels with very specialized customers that will produce SACDs furthermore. But I´m sure that DSD is completely irrelevant for the audio future. The future is PCM not DSD.

I trust you that DSD 128fs is a good thing. I tested it a few years ago and I must admit that this higher DSD rate is quite good. Perhaps not as good as 192/24 Bit PCM but much better than 64fs. This can be explained very easily. One important thing is that the high level quantisation noise begins much higher than at 64fs DSD, now it is more comparable with 192kHz PCM. The only problem is that no consumer format exists for it. And it is not possible to convert 128fs to 64fs without a complete requantisation, and thereby you lose the better quality. This process is unfortunately not comparable with a downsampling in the PCM world. What converters do you use for 32/352,8kHz? I thought it would physically be impossible to design converters with more than 24Bit. Or do you mean mixdown, not conversion? In my mind higher sampling rates than 192 kHz are unnecessary. I´m very convinced that 44,1 kHz or 48kHz is not enough for music recording, but already the step from 96kHz to 192kHz is much more difficult to decide. Higher sampling rates than 96kHz are physically not necessary because music signals contain frequencies till about 35-45kHz. Therefore 96kHz sampling rate with its 48kHz bandwidth is enough. 192kHz gives us a small additional improvement due to the higher precision of the conversion process. Perhaps converters with 8x sampling rates can be better again. But because I reach a fully transparent signal path at 192kHz I think the margin should be very small. In my mind we will get no race about higher and higher sampling rates. This is mainly marketing, not more. In one case I would admit you, if you want to do the mixdown and other processing in the digital domain, too. Then even 352,8kHz is perhaps not really comparable with the precision of good analog audio technics. I tried digital mixing with 192kHz and it is far away from what I want, but perhaps with sampling rate doubled again it becomes good. This is an interesting point, but only interesting for professionals. I´m convinced that we need an audio format higher than 48kHz, but I´m convinced, too, that we don´t need more than 192kHz. I think we should not confuse our customers with other things. Regards, Ralf

We never produced DSD recordings, we tested converters with DSD but the result after AD-DA conversion was very disappointing compared with the analog source and other modern high sampling PCM converters. SACD is the biggest audio nonsense ever developed. Just a few days ago I read a very good article in a German studio magazine. There the technical background of this experience is desribed very good. SACD was a good idea outgoing from the technology of the early 1990th, based on 48kHz/20Bit converter chips with 2,822MHz/1Bit delta-sigma modulators. But with the development of 96k chips it already lost his right to exist. And now compared with modern multibit chips - flagship performance is 6 Bit / 12,288 MHz delta-sigma modulator - the resulting 96kHz or even 192kHz / 24 Bit PCM signal contains much more resolution than SACDs 64x DSD. SACD is not more than 20Bit/48kHz performance with the disadvantage of very high level high frequency quantisation noise and massive noise shaping. And it is much easier to get a high quality signal @96kHz/24Bit out of the 6Bit/12,288MHz stream than 64fs DSD (SACD). Therefore it should be no problem to achieve the SACD sound in a 88,2kHz or 96kHz PCM signal. But: The result would be much better if the signal would never have been a DSD signal before. Straight 96kHz is much better. I personally hate SACD because it kills the music. I can understand why some people like it for its technical clean sound, but at all SACDs I ever heard, the CD layer plays more music than the SACD layer. And I´ve talked to many people, also customers, who said the same. I´m very happy that now the market does not longer need SACD and we now are able to distribute high resolution as downloads in real 96kHz and 192kHz 24 Bit resolution. I would classify HD downloads outgoing from SACD similar to very good CD quality, by no means better. Some labels that have produced SACDs did their recordings in HD PCM. Now they have the advantage over pure DSD recordings, regarding download offers. Best, Ralf Koschnicke ACOUSENCE

Thank you JC for this advice! There actually is a crazy bug with the text links. If you click in the middle of the line, funnily enough just on the word "quality", the link always goes to the 44,1kHz version. The text links of the other words work right. I think this is the explanation for the different feedback. Best regards Ralf Koschnicke ACOUSENCE records

Hi all together,<br /> interesting discussion, I´ve checked the downloaded demo file of Kent Poon´s “Freddie” by myself and I found a hard limit in frequency content at about 23 kHz, all frequency content above is at -90dB and lower. Therefore the original recording was not just done @44,1k. If the 192k version would be only an upsampled CD, this limit must be at 22kHz. In the meantime I saw his post where he explained it and I can confirm, Kent´s explanation is absolutely reliable.<br /> On the other side a limit to 23kHz due to a high cut filter with such a large slew rate and attenuation has nearly the same effect like a resampling to 44,1kHz. That´s theory, but you can also hear it if you compare the files by listening. So I have problems with understanding Kent. I absolutely agree that the filters you need for conversion at lower sampling rates cause big problems. But you get exact the same problems if you manually apply such filters. It remains only the benefit due to better filters in the DA converter. In my mind this is not enough reason for recording and especially selling 2fs or 4fs audio.<br /> <br /> But this discussion gives me a reason for writing “a little bit” more. My experiences are absolutely different. It is not easy for me to repeat a discussion here in English, I had in Germany many times before within the last years. But please let me try to explain some things, because here comes again the old but wrong argument with the ability of human hearing. And also the old discussion that CDs can sound good is not very helpful if we talk about the benefit of recordings with higher sampling rates. <br /> Fact is, CD is a standard and many people are very happy with this. Many people are happy with MP3, too. Also a fact is, that the CD is not able to capture all details in music and sound which are present in the natural performance and e.g. a good analog recording can capture or a good old vinyl record can. <br /> As an audio professional I have to produce CDs for many years and sure it is possible to have much fun with CDs. But as a private music enthusiast I know the great realism and the fantastic musical intensity of good vinyl LPs. Everyone who knows LIVING STEREO or the old VERVE and IMPULSE recordings does understand what I mean. And since we´ve started recording @96k or 192k we are able to capture some of the characteristics I like so much when listening to this records. Sure, it is still not easy to handle 2fs or 4fs really good, however it is very easy to do high sampling recordings that sound not better than a CD. I think this is the reason why so much professionals do argue against high sampling rates. E.g. some professional DAWs have massive timing problems when handling 2fs multitrack. <br /> <br /> Yes Joel, if you observe only one ear, you are right, we can not hear frequencies above 20kHz. But normally we have two ears, and it is proven by science that both ears together have a much better time resolution than it would be possible within the 20kHz bandwidth of one ear. The existence of two transmission channels can not compensate this. Please try a simple experiment: Go with one ear in front of one loudspeaker. Perfect would be a little but very good system or a dual concentric type like Tannoy. Then listen to music, at best a piece with some percussion. Now compare the resolution of your “hear system” when you hear with both ears or when you put your finger into the ear which is on the opposite site of your head (not the ear in front of the speaker). You´ll find that your hearing improves the time resolution of the perceived signal with the sound that reaches the second ear, even due to reflections of the room. E.g. you hear a better structure in percussion or other very complex sounds like piano, strings etc. This is very impressive, our human hearing is such a fantastic apparatus. You can do the same experiment with a clock that ticks very loud. Always the reproduction you perceive with both ears is much more sharp than with only one ear. This fact alone shows very clear that the simplification, done during the development of the CD, is wrong. <br /> But perhaps this is not the most important point. If you want to do a digital recording @44,1kHz, you have to cut off all frequency content above 22,05kHz. This does not only cut down the time resolution of the music recording due to the reduced bandwidth, therefore you need a filter with a large slew rate, too. And this filter does cause a huge amount of time smearing. (Please have a look at image 2 on ths site: http://www.acousence.de/Seiten/digital_cd_en.html) <br /> The very important keyword is the time resolution. Both effects - limited bandwidth and time smearing - cut down the time resolution, and this is not good for the sound and especially the music. <br /> <br /> We at ACOUSENCE offer HD recordings for a few months, as FLAC downloads in 16/44, 24/96 and 24/192 (http://www.linnrecords.com/label-acousence-classics.aspx) and as DVD discs. We also offer some “Comparison Kits”, including the normal Audio-CD. All our productions during the last ten/twelve years are done @96kHz. And all this time I was very sad when I mastered a new CD from the HD master, because the CD format is absolutely not able to capture all the music the master contains. Now finally our customers are able to hear all our recordings in real HD master quality. And many customers wrote us and confirmed the fantastic realism and intensity especially of the 192kHz files. The true benefit of recordings at high sampling rates is the considerably improved realism of sounds and the capture of finest musical details as well as the atmosphere of an artistic performance. For me the last two topics are much more important than the sound. Sure, it is absolutely fascinating what realism now is possible, but I think for recognizing the real profit you need a very good playback system. Contrary to the intensity of the musical performance remains noticeable all the time. And we are talking about music, not about a technical system for transmitting sound pressure waves. Now the musicality of digital recordings becomes comparable with a very good analog recording and – most important – the real artistic performance. (Perhaps you could find more interesting information at this site respectively the two links there: http://www.acousence.de/Seiten/analog_lp_en.html)<br /> <br /> On this site http://www.acousence.de/Seiten/technology.html we describe very clear our production process. Our offered HD recordings are no fakes. All is multitrack recorded at 96k, mixed analog (very important!) and then recorded at 192k (Even a 96k master is not able to capture all details of the complex analog stereo mix of those many 96k tracks). The offered 192k files contain an exact copy of the output of our mixing desk with 24/192 AD conversion. Then all lower quality files are the result of a conversion process outgoing from this 24Bit/192kHz master file.<br /> <br /> I think it is very important to say this, because our music industry has done so much mistakes during the last decades. With the upcoming “listening from file formats” we get a so wonderful chance for a real evolution in music recording. And therefore I hope that not too many companies do destroy this chance due to hoaxing their customers. Many companies feel that they are able to sell with the attribute “HD”, “HR” or “192kHz”, but they nearly all have the problem that they have no existing recordings done at higher sampling rates. Perhaps they have problems with understanding why they should do this, too.<br /> <br /> Kent Poon has his own reason for doing it like he did, and I respect this, that´s absolutely okay. But I´ve checked other very well known 4fs HD recordings and I found similar things without any declaration. After reading this story by Juergen, now I fear bit by bit that 192k becomes more and more a virtual argument for selling recordings, but without the real benefit which is possible. Unfortunately this delivers a very good argument for all people which have ever known that high sampling rates are nonsense. And in the past we had a similar situation with DVD-A and SACD and we all know the result. I don´t want to live in a future with MP3 as THE standard audio format.<br /> <br /> Best wishes,<br /> Ralf Koschnicke<br /> ACOUSENCE records<br />

Hi, please try this free demofiles: http://www.acousence.de/Seiten/digital_cd_demo_en.html No 24/176.4, but also quadspeed: 24/192. Our recordings are available at LINN records: http://www.linnrecords.com/label-acousence-classics.aspx Best wishes Ralf Koschnicke ACOUSENCE records