161–170 of 445 posts
I've said it before, this video demo is one of the very best I've ever seen. https://xiph.org/video/vid2.shtml So well prepared, so well presented, so little that could be removed without ruining it. I aspire to do such good demos but always fall so short.
Makes me long for the day when consumer audio products were proudly labeled as having a 1-bit DAC. Nowadays you couldn't sell audio hardware that made such truthful statements about its internals. Gotta have the bits.
If the only thing you do with your music is listen, then yes, 24/192 delivers questionable value compared to the more popular 16/44.1 or 16/48 formats. However, all musicians I know use these high-rez formats internally. The reason for that, when you apply audio effects, especially complex VST ones, these discretization artifacts noticeably decrease the result quality. Maybe, the musicians who distribute their music…
I do not believe it. 24 bits is definitely needed for processing (better yet, use floating-point). Not 192 kHz; no friggin' way. Repeated processing through multiple blocks at a given sample rate does not produce cumulative discretization problems in the time domain; it's just not how the math works.
That isn't entirely true. e.g. It's common for an audio DSP to use fixed point 24bit coefficients for an FIR filter. If you're trying to implement a filter at low frequency then there can be significant error due to coefficient precision, that error is reduced by increasing the sampling rate.
and yet, we are nowhere near being able to electronically reproduce a live acoustic music performance. have you ever walked by a bunch of musical sound coming out of a room and thought to yourself, "wow, those live musicians sound great" only to discover it was just a stereo playing? nope. as engineers we will never solve this problem as long as the "44.1kHz is good enough" dogma is perpetuated. here's a question. wh…
In my own experience, the vast majority of consumer audio gear doesn't even take full advantage of 16/44.1. I've made live recordings of my classical chorus, and if I want it to be listenable on typical consumer gear, I have to apply a significant amount of dynamic range compression - otherwise, it will be either too quiet, or it won't handle the peaks when played at concert volume. That being said, I'm using quite a…
Good points in the article, but it has some flaws. The problem whenever somebody writes about digital audio, is that it is very tempting to hold on to sampling theory (Nyquist limit, etc) and totally discard the problems of implementing an actual Analog-Digital and Digital-Analog chain that works perfectly at 44100Hz sample rate. I agree with the assesment that 16 bit depth is good enough; even 14 bit is good enough…
There is no reason the DAC or ADC can't internally do the resampling itself. In fact, most take it to the extreme - they resample to several MHz, but at 1 bit per sample, with noise shaped dithering as explained in the video linked elsewhere. They then have a very simple analog low pass filter - the noise is shaped far into the stop band of the filter. That is no reason to store or even process your music at higher r…
You are describing alternative (b) i mentioned above: digital filtering plus oversampling. This also isn't without side effects.
Oversampling a 44KHz signal is not the same as having a 192Khz material to start with. Very different.
If the only thing you do with your music is listen, then yes, 24/192 delivers questionable value compared to the more popular 16/44.1 or 16/48 formats. However, all musicians I know use these high-rez formats internally. The reason for that, when you apply audio effects, especially complex VST ones, these discretization artifacts noticeably decrease the result quality. Maybe, the musicians who distribute their music…
I do not believe it. 24 bits is definitely needed for processing (better yet, use floating-point). Not 192 kHz; no friggin' way. Repeated processing through multiple blocks at a given sample rate does not produce cumulative discretization problems in the time domain; it's just not how the math works.
Both your inputs (ADC) and outputs (DAC) are fixed-point. Why would you want to use a floating point in between? Technically, 64-bit floating point format would be enough for precision. But that would inflate both bandwidth and CPU requirements for no value. 32-bit floating point ain’t enough. Many people in the industry already use 32-bit integers for these samples.
> Not 192 kHz; no friggin' way.
I think you’re underestimating the complexity of modern musician-targeted VST effects. Take a look: https://www.youtube.com/watch?v=-AGGl5R1vtY I’m not an expert, i.e. I’ve never programmed that kind of software. But I’m positive such effects are overwhelmingly more complex than just multiply+add these sample values. Therefore, extra temporal resolution helps.
BTW, professionals use 24bit/192kHz audio interfaces for decades already. E.g. ESI Juli@ was released in 2004, and that was very affordable device back then.
That explains why some people can tell the difference between 44.1 khz and the higher-resolution sampling rates. It also means that the ideal audiophile sampling rate is somewhere between 58khz and 60khz, not 192khz.
and yet, we are nowhere near being able to electronically reproduce a live acoustic music performance. have you ever walked by a bunch of musical sound coming out of a room and thought to yourself, "wow, those live musicians sound great" only to discover it was just a stereo playing? nope. as engineers we will never solve this problem as long as the "44.1kHz is good enough" dogma is perpetuated. here's a question. wh…
Good points in the article, but it has some flaws. The problem whenever somebody writes about digital audio, is that it is very tempting to hold on to sampling theory (Nyquist limit, etc) and totally discard the problems of implementing an actual Analog-Digital and Digital-Analog chain that works perfectly at 44100Hz sample rate. I agree with the assesment that 16 bit depth is good enough; even 14 bit is good enough…
I don't think there are flaws in the article as you claim. It already explicitly explains that oversampling at the ADC or DAC is an acceptable engineering solution: > Oversampling is simple and clever. You may recall from my A Digital Media Primer for Geeks that high sampling rates provide a great deal more space between the highest frequency audio we care about (20kHz) and the Nyquist frequency (half the sampling ra…
Of course it is acceptable. Even 14 bit audio at 36KHz with a great DAC would be fairly nice, acceptable.
What the article claims is that 192KHz is useless, of no benefit. And i contend that it is of benefit when you want more than just good or acceptable performance. Not if you have a run of the mill DAC and OK headphones/speakers, but it is if you are a music lover and critical listener.