Just placed into my system an Ayon S3 streamer/ server. This has the ability to upsample 16/44 to 24/192 on the fly.
So last night was comparing the effects on some well known to myself music and this is what I heard whether it can be measured or not.
At 24/192 the music was smoothed out for want of better description. Some of the common CD glare and harshness was tamed, really noticeable on cymbals. For me I preferred the majority of the 16/44 I upsampled to 24/192. A few better recorded tracks did not seem to benefit although the change did not seem to be detrimental either. YMMV
mzkmxcv374 posts Why don't you try yourself, instead of asking stupid questions and making stupid comments,and find out - as clearly you have either .. a - not tried or b - have cloth ears or c Have rubbish equipment or maybe as I believe all 3 Please report back as clearly your findings and opinions are so accurate
If I listen to a Jaguar engine when it is away from me in a garage and compare it to the sound if I was standing next to it, would I hear a difference? It is the same sound isn't it ... so according to you must sound the same
@uberwaltz congrats on your new Ayon and your experience sounds about right upsampling can help some recordings and not others. As with all recordings the initial mastering/recording quality matters most.
My experience is that the worse recordings are improved more than the better recordings - which seems logical to me. As you say, a good recording is a good recording, regardless. A poor recording of Band of Gold - Freda Payne - gets some sort of life in it, but no matter what, I find nothing can bring Ultravox to any life at all! Were their recordings known to be poor?
uberwaltz4,443 posts Yes I have same impression. Tried many times to enjoy upsampling to dsd, but all flavour and air dried up and sounded very mechanical. On the basis that most of the time 192 worked for me best (sometimes 96 suprisingly) I took the view to fix at 192 and just enjoy as much as possible without always wondering ... what if I went down to 96!.. Exactly the view of my long term dealer, who's ears and opinion are usually spot on
Thats a really poor analogy. Of course it won’t sound the same, as you are changing the physical environment, so you now are dealing with altered reflections/reverberations/echo.
To paraphase, 'its the source material stoopid'. Even supposedly top mastering studios like Bernie Grundman still churn out crap sounding CD's, its a disgrace really.
I admit, I did not read all the personal opinions (which they are). I would every second, hour, day and week prefer a well built, state of the art cd/sacd player that does not upsample to DACs market to the "highest is best" crowd. To paraphrase the old saying, "if the first watt is bad the next 500 will be just as bad". And to quote wikipedia,
"A commonly seen measure of sampling is S/s, which stands for "samples per second". As an example, 1 MS/s is one million samples per second.
When it is necessary to capture audio covering the entire 20–20,000 Hz range ofhuman hearing,[5]such as when recording music or many types of acoustic events, audio waveforms are typically sampled at 44.1 kHz (CD), 48 kHz, 88.2 kHz, or 96 kHz.[6]The approximately double-rate requirement is a consequence of theNyquist theorem. Sampling rates higher than about 50 kHz to 60 kHz cannot supply more usable information for human listeners. Earlyprofessional audioequipment manufacturers chose sampling rates in the region of 50 kHz for this reason.
There has been an industry trend towards sampling rates well beyond the basic requirements: such as 96 kHz and even 192 kHz[7]This is in contrast with laboratory experiments, which have failed to show thatultrasonicfrequencies are audible to human observers. In fact, in some cases, ultrasonic sounds do interact with and modulate the audible part of the frequency spectrum (intermodulation distortion),degradingthe fidelity.[8]One advantage of higher sampling rates is that they can relax the low-pass filter design requirements forADCsandDACs, but with modern oversamplingsigma-delta convertersthis advantage is less important.
TheAudio Engineering Societyrecommends 48 kHz sampling rate for most applications but gives recognition to 44.1 kHz forCompact Disc(CD) and other consumer uses, 32 kHz for transmission-related applications, and 96 kHz for higher bandwidth or relaxedanti-aliasing filtering.[9]Both Lavry Engineering and J. Robert Stuart state that the ideal sampling rate would be about 60 kHz, but since this is not a standard frequency, recommend 88.2 or 96 kHz for recording purposes."
In other words, higher is not necessarily better from this armchair.
Folks, I don't have time to go into all of it now, but there are LOTS of really serious errors in the statements being made here. e.g. "thinning out the music", e.g.: "you cant hear above 20 kHz so why 192?" These make really critical mis-assumptions about what is going on in up sampling. First, go learn about up-sampling and interpolation filters. Then learn about reconstruction filters and their issues. Then think about how much better you make things if you first interpolate and then feed it to the reconstruction filter. Lots of analog issues get much easier. It need not change the original data one bit (both literally and figuratively) Bottom line: this is all about making the job of the end analog filters easier and less likely to produce artifacts. And done right it works terrifically well. Done wrong, all bets are off. Don't do it wrong :-)
Quote: (partial) "There has been an industry trend towards sampling rates well beyond the basic requirements: such as 96 kHz and even 192 kHz[7] This is in contrast with laboratory experiments, which have failed to show that ultrasonic frequencies are audible to human observers. " Comment: Whoa! None of this is done to reproduce ultrasonic frequencies. In fact the analog filters remove them as best they can. It is done so that the analog filters can be lower-order and more phase-correct, and/or to allow them to BETTER remove the ultrasonic frequencies (which exist, whether you like it or not in a stepped eave out of a DAC) more completely. Heck, read the app notes from any major DAC provider like TI/Burr Brown and see the noise after their reconstruction filters. not so good you will find. But with proper up sampling, we move the noise higher making it easier to filter. It is these partial truths that continue the crazy arguments between engineers trying to make better sound and audiophiles that are being a) confused and b) sold stuff that does not sound as good as it otherwise might. Rant off.
These make really critical mis-assumptions about what is going on in up sampling. First, go learn about up-sampling and interpolation filters. Then learn about reconstruction filters and their issues. Then think about how much better you make things if you first interpolate and then feed it to the reconstruction filter. Lots of analog issues get much easier. It need not change the original data one bit (both literally and figuratively)
Look at the measurements for any non-MQA compatible DAC, all their filters shouldn’t attenuate any frequencies you can hear, especially from companies such as Chord, and any phase errors are going to be in the treble where it won’t be audible on any decent DAC.
Also, if you are upsampling to a non-multiple (44.1 to 192) it no longer is bit-perfect, and the rounding errors would need to be minimized by the DAC/software.
I’ve yet to see any actual benefit to upsampling when disregarding any limitations by the DAC used.
Wow, this is getting out of hand in breadth and how fast it progresses. Someone commented on what i said (quoted me) and then went on to speak of Chord DACs having the best filtering they knew of. It **appeared** that they were using it to counter my argument that up/over sampling helps with the filtering. Let me be clear: Chord over/up samples. Their data stream runs at 104 mHz/mbps - far, far, far above the raw output of redbook ( around 1.5 mbps). Their sound is not all due to filtering either. They pay special attention to timing and jitter (as one must, since the reconstruction filter integrates over the sample value and the time). They further use a different method to get the analog pulses (flip-flops, let's not go down that art-hole) that has some claimed advantages. But it in NO WAY counters what I wrote. But with the loose writing here it difficult to even understand what point people are trying to make. G
I know the Chord upsamples, so does the Benchmark, and so do many others, I’m specifically referring to upsampling before the DAC. DACs do this as it helps with jitter, doing this before the DAC doesn’t help.
mzkmxcv381 posts It is abundantly clear that this chap has NOT experienced upsampling and is spouting verbage about measurements he does not understand anyway and he gains satisfaction from writing and seeing his own words in print that have NO BENEFIT TO ANYONE. Can he please desist from irritating sensible people who DO know what they are talking about and have EXTENSIVE listening experiences of the benefits. Can we have a rule on this forum that if you have not tried DO NOT MAKE SPURIOUS comments on those who HAVE, disputing their ears and findings.
" know the Chord upsamples, so does the Benchmark, and so do many others,
I’m specifically referring to upsampling before the DAC. DACs do this
as it helps with jitter, doing this before the DAC doesn’t help." No, they do it to interpolate and reduce the challenge for the reconstruction filter. Again, don't believe me, read a Burr-Brown application note if you prefer. There was also a point about up sampling being no longer bit perfect. If you understand interpolation, which is a distinction without a difference. If you understand interpolation, you know that it creates new data points which a) raise the frequency of the noise to be filtered and b) invent a new intermediate level - beyond 16 bits -- which in effect raises the resolution. It is absolutely true that no new data exists and it is absolutely true that in a world of perfect, phase-coherent, 200 dB/octave filters, it would be un-necessary. But we are not in that world. So, at 88.2 kHz we get effectively 17 bits. at 176.whatever we get 18 bits. I acknowledge that at 96 we do nto have a simple multiple, BUT -- HUGE BUT -- all the original data points remain adn can be reconstructed so in practice it is bit perfect. Its like saying "i used to have $100, this crook gave me $2 and now I don't have $100 anymore". Uh, true, you have $102. You are free to throw $2 on the ground - and the DAC is free to toss any bits on the ground; but it will use them to make its later job easier. And yes, there are differences pre-DAC and in-DAC but they are ALL BEFORE the actual conversion - either via PDM (sigma-delta) or PAM (ladder).
Using a multiple to upsample is an easier job and hard to mess up. Using a non-multiple helps with jitter but is harder to implement.
As for up/over sampling before the DAC, it matters whether either side takes care of intersample overs, the Benchmark models for instance does take this into account and lowers the level before oversampling.
Interpolation does “destroy” the original data while synchronous upsampling preserves the original data. It matters how good the interpolation is. Going with your currency analogy, it would be like if I had a $100 USD but someone took it and gave me $100 in Bahamian currency, which averages a 1:1 conversion with USD, but sometimes it’s not, it has been worth 0.98:1 and sometimes 1.02:1.
Stereophile’s measurements of DACs shows how much aliasing occurs by using a high frequency test tone, if you look at the AudioQuest DragonFly Red’s measurements for instance, the aliasing image at 25kHz is below -100dB.
I can’t comment on phase though as I haven’t seen that tested before, but we are very forgiving of phase errors (not talking phase mismatch) in the treble that I doubt any decent DAC will have an issue.
Snopes is a very reliable source for debunking urban legends such as the green marking pen used on CD's. Where's the evidence they are unreliable? ... thought so.
Cables can sound different in different systems, but above a certain level of materials and build quality you are hearing ONLY differences. They may sound small to some and significant to others. A $5000 power cable DOES NOT sound better than a $500 power cable. This is even more true with USB and other cables carrying a digital signal. Super expensive cables often look really cool, but you're kind of getting ripped off once you get into the four and five figure pricing. If you can afford it and think you like the resulting changes it makes to your system then just rock on!
As Stereophile has reported when testing high-end 16bit CD players and DACS, either technology, 16b or 24b, or a 44k sampling rate or much higher like a 196k rate can both sound fantastic when implemented in well designed gear. It is not at all a given that the higher sampling rates always sound better. So no, more is not always better.
Personally I prefer the sound of vinyl records and all analog playback over digital despite some of the advantages of the latter. It's just a personal thing, but vinyl sounds more "musical" to me.
There is a LOT of snake oil being sold to audiophiles and there is little doubt in my mind that many of these products rely more on psychology than they do science. Your mind will change what you hear based upon your own confirmation bias-- and of this there is no doubt.
Every system will sound different based upon the combination of gear and the room that it's in. Your room will have a vastly greater impact on the sound of your system than anything achieved changing to more expensive cabling-- assuming you are already using something decent.
My phono cartridge will have a much greater impact on the sound of my system than my amplifier. So will my speakers. So will my phono preamp. Digital gear has improved immensely over the last fifteen years or so regardless of the bit rate. Does that mean I'd prefer a lower bit rate? Nope. Just saying...
Where is the evidence Snopes is nothing more than a bunch of mud-raking know-nothings? What would they know about the Green Pen? I’m guessing about as much as you do. No offense to you personally.
I spent a lot of time using Adobe Audition and evaluating what was the best conversion of my 33/LP album collection to MP3s. I've been doing it now for more than a decade and ended up with Audition capturing at 32-bit/44Khz Windows-compatible wave files. I also edit in 32-bit. When the WAV file is complete, I use LAME to convert directly from the WAV file to MP3 at 128/variable. The result is roughly about 1Mb per minute of MP3 music file(s). I have had multiple people say the MP3s are the best they have ever heard. (from LP) Audition can 'Analyze' the WAV and the resulting MP3, and the resulting dynamic range is usually better than CDs, even though the source is from LPs. With that result, I have not changed my conversion process in more than a decade, as I see no reason to change it. Please don't flame me, maybe it can be done better, this just how I did it with modest gear. (I am not Bill Gates)
the resulting dynamic range is usually better than CDs, even though the source is from LPs.
Vinyl is reported to have a noise floor of roughly -65dBFS, so I don’t see how any ADC or conversion technique can make it better than CD at -96dBFS, unless you are adding noise-shaped dither, in which case the same can be done for 16/44.1.
The Ultravox tracks 'Vienna' and 'All Stood Still' sound good to my ears, so my answer to your question is "No" Ultravox do not make poor recordings. These are the only two tracks I have that are digital. My personal views only.
I did my best to evaluate precisely that. The WAV file being the reference, (which included click/noise reduction) I experimented with many iterations of LAME/MP3 settings until I could not tell the difference in the conversion. I don't consider my ears the best, as they've been abused/damaged in the workplace. So these 'other' ears are a valuable resource for this result. And I edit in 32-bit on purpose, not 16.
Vinyl is reported to have a noise floor of roughly -65dBFS, so I don’t see how any ADC or conversion technique can make it better than CD at -96dBFS, unless you are adding noise-shaped dither, in which case the same can be done for 16/44.1.
I submit you may not have experienced the range of features and options available in Adobe Audition. Adobe purchased Cool Edit 2000, and has made a lot of improvements. (I started with Cool Edit 2000)
Audition's 'Analysis' of LPs before click and pop and other noise reduction says LP dynamic range starts in the 40s for bad ones, and I'll agree with your number of 65 for only the very best. (very rare) I can't recall the last time I saw one that good. They are typically in the 50s. (per Audition)
Just to mention that the figures touted for CD SNR and Dynamic Range of 90 dB or more are strictly theoretical. In practice the ordinary drawbacks of the playback system and room not to mention the *intentional overly aggressive dynamic range compression* that’s fairly rampant in the industry over the past twenty years or so obviously limits those numbers. How much? Answer at 11.
The objectively BEST sample rate for playback ... drum roll please ...
Is the one used in the mastering room
On the approved recording. There is no gain to moving from there, and with something like MQA that adds distortion, there is a loss in quality. Having said that as #1 the next factor in listening is your room. Most rooms are not good, and sure we can put $20,000 or $200,000 in gear in a bad room ... but it’s rather silly if you understand the room to be over 50% of the sound from a pair of speakers. And if you doubt that figure, set up the hifi in the garage or basement.
but it’s rather silly if you understand the room to be over 50% of the
sound from a pair of speakers. And if you doubt that figure, set up the
hifi in the garage or basement.
I would honestly say this is conservative. I've measured rooms, and with +- 20 dB nulls and peaks, a room can have tremendous effects on the output power in the bass, the timber, and the ability to resolve spacial cues.
It is also true that we can train our ears to reduce some of these effects. Try recording a person's voice, or just your stereo in a room, then listen to it with headphones. You'll be amazed. With training, you can teach yourself to stop blocking this, and you suddenly can hear the room itself.
This thread is about high sampling rates as a means to improve AtoD/DtoA because CD sounded crap. Now you can blame the medium, the mastering, the sampling rate or the playback system but not room acoustics.
CD already offers us better quality than we can hear in a room, so that argument doesn’t hold.
I also only skimmed the article, but I didn’t see any mention of ADC/DAC, it’s merely saying your hardware likely can’t handle the higher sampling rate, the Benchmark DAC3 for instance acruslly downsamples 192kHz audio to 110kHz, as the chips used have poorer performance at 192kHz, and not that you’d hear the difference of 110 vs 192.
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