261–270 of 274 posts
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you might want to see if your DAC re-clocks incoming optical, if not then it's relying on the cheap clock generator from your computer
That cheap clock source is good enough for 16 GHz PCIE signaling, but Im sure you will be able to hear the jitter!
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Am486DX/100 was enough to decode and listen an MP3 at 22KHz (and maybe mono?) and was more than enough to listen for 44/16/2 PCM. It's 31 y.o. today.
I remember playing 44khz 16-bit stereo MP3s encoded at 128 kbit/sec on a 133 Mhz 486. It gobbled like 90% of the CPU and I had to make sure I gave it a pretty large buffer so it didn't stutter when an app claimed CPU for more than a second, but it worked.
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Yeaaaaaaaaaahhhh.... you might be a tiny bit deaf. OTOH, we know nothing of your audio equipment nor how its setup.
HiFiMan Sundara headphones, focusrite scarlett 2i2 interface. Although, it turns out the Scarlett is set to 48KHz anyway, and I can't seem to change it easily under linux. Not that it seems to matter for my ears, lol. EDIT: Did some more ABX testing with a CD-quality track that I'm much more familiar with ('Introduction' from the Mirrors Edge soundtrack, which has been my go-to for comparing audio gear for the last d…
At 48khz, with a tone generator (not rando Youtube videos or anything), you should be able to clearly hear up to around 16khz (as in, can tell pitch of tone), and be able to tell a tone is being played at all up through 18khz, and hopefully up to 20khz.
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It's just means you have a shitty audio tract with not enough shielding. Move to SPDIF/TOSLINK.
I have an external audio card, if I put it on a laptop I can hear the modem-like sounds. I wonder why it is so sensitive, should not DAC produce strong signal that cannot be easily affected by radio waves? Also my headphones are extremely sensitive. I can touch the ring and sleeve of a jack with a finger, and touch a metal bed frame with a tip and I hear quiet clicks as I move the tip along the metal. Sometimes I do…
And this is not the case of 'stronger than', it's 'strong enough to be caught up by anything resembling an antenna'.
You hear the interference because some analogue tract of your system:
Do transmit the radio waves everywhere
Does receive the inference from the all electric things around.
And now I'm skipping 'cuz I'm inna a bar and it's more interesting.
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> 24/192 is also great for digital synthesizers--if you're generating a waveform like a sawtooth that has theoretically instantaneous transitions, they can eat as much frequency as you can give them. So if your synthesizers do not use proper band-limited oscillators then 192KHz is _FAR_ too slow. You'd want to be running at hundreds of KHz, perhaps a few MHz. In reality synth software that doesn't sound like crap use…
> I think given how fast computers have become relative to digital audio there is probably a good case to just make any "modular synth" run at 32-bit 480KHz or even 4.8MHz through every stage that could process the audio. 1. It should run at FP64 if you want to preserve filter resonances, etc. 2. At 10x/100x fixed-rate oversampling, even a modern "fast" CPU will have very few cycles per (higher-rate) sample to run th…
For adaptive rate I think the issue there is you have a hard-realtime constraint for this usage (even if you wouldn't mind rendering offline, you kinda have to hear it realtime to tweak it-- after all you might tweak it in a way that brings out an artifact you like and then be disappointed by the render). Also in the case of a whole modular system having all sorts of different parts needing to be part of the adaptation loop seems pretty hard to me.
My thinking was just in general that 192k is really not enough to prevent aliasy algorithms from messing up. If you are alias safe you can probably run at 48k and be fine. If you're not, you really want to go much higher.
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Yes, perhaps the amplitude was subtly different. This was supposed to be running the DACs to match the source configuration, not resampling into some common format. I think that is an unavoidable part of the whole end-to-end ABX test concept. Maybe it would be interesting to up-sample back into 24/192 and play both in that mode. But then people would argue about what type of up-sample to use. I was in my mid 20s for…
Speaking about up-sampling, Im curious to know your opinion on the benefits of it. Im sending CD resolution audio as well as web streams from soundcloud.com to cambridge audio azur 840C and its not clear if its the up-sampling that makes the difference or their per channel wolfson dac arch. The iPod Video with their dac sounded amazing with just normal AAC files compared to the iPods before or after it.
Second guessing it by upsampling in front of it seems dubious to me. It might help in some cases where the DAC designers were thinking of different objectives or just didn't do a great job. It might also help with some other issues, like if the dac is timed off the input clock and the input clock sucks and the upsampler retimes the signal.
Of course the upsampler designers could also get it wrong, be aliasing the hell out of the results, and happen to like the sound of the corrupted audio. :P
The effects are all objectively measurable however-- with expensive equipment at least. I think I'd want to set test results with a particular hardware combination before sticking an upsampler in it. OTOH, if there already was one there because it's just some built in feature of some kit I wanted to use otherwise, I wouldn't worry much about it. Particularly if that kit has been reviewed by people with proper test gear and they didn't decide that it was broken.
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Got it. Grammy voters love Collier's mixes. What about Tony Maserati? He can clearly tell the difference between 44.1 and 88.2. If your argument is that these engineers can't hear the difference - you are going to be disappointed. They can. Even Dave Pensado who mixes at 16/44.1, does that because he rejects the idea, he can hear the difference according to him.
I can find no evidence that Maserati (or 99% of any other mix or mastering engineers) has ever tested his "appreciation" of the "crunch" at 44.1 in a double blind environment. It is always amazing how much that is claimed about what people can hear fails to show up when tested in this, the only acceptable scientific way. Perhaps Maserati has done this, and could still tell the difference. In which case, he should car…
And since there are no double-blind studies supporting this tech, both using and adding any of these features to your software would only be propagating this scam further? I.e. far more than just lecturing "about stuff that isn't true" this actually physically implements features that are not true?.. OK, at least you are staying consistent.
(Also, looking forward to you discovering that there are not many double-blind studies supporting the "delusion" that the effect of a low pass filter is real and not just something we can measure but can't hear.)
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Can you give any examples of people identifying these artifacts in a/b tests? Who has the best ears? What can they detect?
OK, so we are entering the stage of "can you provide a double-blind study link". I can look it up, I am not a researcher. Here is one: https://qmro.qmul.ac.uk/xmlui/bitstream/handle/123456789/134... I know from my 20-ish year mixing experience that I can hear the difference when mixing. Is it good evidence? No. So we can agree to disagree then.
I'm not disagreeing with you. I'm really curious about the limits of what people can hear, what can be taught and what is rare.
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Decades ago, I was treated to an ABX test in my brother's recording studio. I easily recognized and preferred a 24/192 master he played versus the 16/44.1 down-mix. I honestly don't know whether there was something wrong with the down-mix, but qualitatively it did feel like it was "muffled" and coming from speakers, while the master really felt like live performance. He was surprised that I could tell them apart. I a…
> Decades ago, I was treated to an ABX test in my brother's recording studio. I easily recognized and preferred a 24/192 master he played versus the 16/44.1 down-mix. I honestly don't know whether there was something wrong with the down-mix, but qualitatively it did feel like it was "muffled" and coming from speakers, while the master really felt like live performance. He was surprised that I could tell them apart. A…
No one can hear the difference between properly mastered high res files. I will happily put money on it.
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> I think given how fast computers have become relative to digital audio there is probably a good case to just make any "modular synth" run at 32-bit 480KHz or even 4.8MHz through every stage that could process the audio. 1. It should run at FP64 if you want to preserve filter resonances, etc. 2. At 10x/100x fixed-rate oversampling, even a modern "fast" CPU will have very few cycles per (higher-rate) sample to run th…
so an 8-core zen4 should be able to sustain more than 300 gflops of 64-bit multiply-adds. At 480khz that's 625k operations per sample. I'll grant the 100x oversampling was probably too ambitious. :P For adaptive rate I think the issue there is you have a hard-realtime constraint for this usage (even if you wouldn't mind rendering offline, you kinda have to hear it realtime to tweak it-- after all you might tweak it i…
These simulations are single core to avoid core-to-core latency. Number of cores isn't relevant unless you want to run independent voices/channels and sum them at the end.
So you start with a very optimistic ~90 GFLOPs of 64-bit FMA on Zen 4. Unfortunately, not all operations are clean multiply-adds. Realistically, you'll need trigonometric functions and LUTs, which are quite slower. Btw, the tradeoff between when to compute vs LUT is very fragile and can change due to a ton of factors (notably integrator algorithm).
Then the data you are operating on won't fit cleanly in AVX-512 registers, requiring spills to L1 cache. Ok, still fast on a modern core.
Of course, the peak theoretical number assumes clean vectorization with double-pumped AVX-512... which also won't happen in practice. Classical DSP will fare better (https://www.youtube.com/watch?v=Ssq0a-YdamM) but SPICE integrators are inherently branchy and divergent. Especially for adaptive integrators, you'll waste a lot of operations trying to "lock in" at the exact time point where the waveform turns a corner. Apple Silicon is better at this messy, branchy code.
So yeah, it's possible-but-hard to hit hard-realtime under these conditions.