[sander99]

Is waveshape the first thing that is analog instead of digital? Or is it how air moves, at the very end of the signal chain? I'm wondering. And since I am definitely not talented in DSP and electronics, I'm curious if someone could "stress" a DAC by using an all pass filter (?), like, adding more time information for whatever reason. I'm afraid the idea attracts me because I know nothing about DACs.

Yes, the waveshape is analogue and it contains information about frequency, amplitude and phase. Even if we start with frequencies limited to 23 kHz, the combination of all those informations into the structure we call the waveshape will result in frequencies much higher than 23 kHz. If we sample it at 44.1 kHz, for example, a significant amount of information encoded in the wave shape will be missing when digital is converted back to analogue. This produces distorsion.

A waveshape is simply the shape or form of a periodical function depicted as a graph.
An analog audio signal is (in most cases) a voltage changing over time.
(And if a microphone is subjected to sound, it is actually subjected to air pressure changing over time, and a graph depicting the air pressure as a function of time would be "analogues" to the graph depicting the output voltage of the microphone as a function of time, so it would have the same or similar shape.)

If you depict it as a 2 dimensional graph with time on one axis and voltage on the other axis that graph has a shape: the waveshape.
Examples of periodical waveshapes a sine wave, a square wave, a triangle wave.
In a way you could also call a complete recording one big complicated waveshape.

Important to know is that every waveshape can be decomposed in (or written as the summation of) a finite or infinite number of sine and/or cosine waves.
(Discovered and proved - for functions in general, maybe he didn't think about sound at all - by a mathematician called Fourier 2 centuries ago!)
If we say a recording contains certain frequencies it means that the decomposition into sine and/or cosine waves contains sine and/or cosine waves at those frequencies.
For example a perfect square wave (that does not exist in the real world) is the sum of an infinite number of sine waves:
sin(x) + 1/3sin(3x) + 1/5sin(5x) + ...
If you listen to a square wave with a fundamental frequency of 10 kHz, the next harmonic will be at 30 kHz, the next at 90 50 kHz etc.
Inside the inner ear are hairs "tuned" to small frequency bands. For some hairs that small frequency band will include 10 kHz. They will resonate with the 10 kHz fundamental and as a result neurons will fire to signal to the brain that something happens at 10 kHz.
There are no hairs tuned to frequency bands that include 30 kHz, or 90 50 kHz, etc.
So you won't be able to hear the difference between a sine wave at 10 kHz and a square wave at 10 kHz.

The sine/cosine decomposition of a band limited signal (with 20 kHz upper limit) will only contain sine/cosine waves below 20 kHz.

The sampling theorem is a proven mathematical statement. What is proven is that if you have samples of the signal at least every half cycle time of the highest frequency in the signal then you can completely and precisely reconstruct that signal. All sine/cosine waves of the decomposotion can be reconstructed including the exact phase and timing! That is if the samples are exact. If the samples only have a limited precision, for example 16 bit, then the reconstruction will contain added noise. (However, 16 bits is enough to keep the noise far below audible levels, assuming you don't listen at such a high volume that the loudest possible parts of the recording will destroy your hearing.)

If you listen to a square wave with a fundamental frequency of 3 kHz there will be some hairs reacting to 3 kHz, some to 9 kHz, if you are lucky and not to old some to 15 kHz, and that's it.
If this signal is recorded and played back digitally with 44.1 kHz sampling rate then the 3, 9 and 15 kHz will be perfectly reconstructed, with the correct timing/phase.
If you listen to this recording you will not be able to distinguish it from the original square wave because all the audible components of the square wave are there (and the same hairs in your inner ear will be reacting).
(That is: unless your amp and speaker will cause audible distortion when fed with the original square wave, which is very likely because no amp can change it's output voltage infinitely fast, and no speaker membrane can move infinitely fast. In this case using the digital recording - or low pass filtering the original square wave - would lead to a audibly better reproduction of the square wave.)

[Edit: corrected a few wrong numbers...]

 

[71 dB]

Clearly audiokangaroo is not interested of learning anything, but instead tries to convince us about the "inferiority" of digital sound and strange ideas about what humans can hear. I made a few posts with patience, but at this point this patience is exhausted. I won't waste my time more. I am done with trying to explain to vinyl users that the reason why their CDs sound crap isn't because the format is inferior, but because crappy sound was for whatever reason put on the CD.

Why don't music producers add simulated vinyl distortions on CD masters on certain type of music genres sounding "better" on vinyl? People would LOVE the sound! On CD!

 

[castleofargh]

I think that vynil is not band limited like digital. It can reproduce frequencies higher than 15 kHz, but with a very low amplitude that is inaudible. Thus is has a very good ability to follow the waveshape. We should admit that in spite of the important amount of distortion a vynil disc can sound very good, and globally better than a CD.

Sorry, no. We shouldn't admit that worst fidelity is better.

I share the opinion of 71 dB, and would add that it's not a topic on vinyl. Feel free to continue this in a proper thread, or make one if you care to discuss this, thank you.

 

[audiokangaroo]

A waveshape is simply the shape or form of a periodical function depicted as a graph.
An analog audio signal is (in most cases) a voltage changing over time.
(And if a microphone is subjected to sound, it is actually subjected to air pressure changing over time, and a graph depicting the air pressure as a function of time would be "analogues" to the graph depicting the output voltage of the microphone as a function of time, so it would have the same or similar shape.)

If you depict it as a 2 dimensional graph with time on one axis and voltage on the other axis that graph has a shape: the waveshape.
Examples of periodical waveshapes a sine wave, a square wave, a triangle wave.
In a way you could also call a complete recording one big complicated waveshape.

Important to know is that every waveshape can be decomposed in (or written as the summation of) a finite or infinite number of sine and/or cosine waves.
(Discovered and proved - for functions in general, maybe he didn't think about sound at all - by a mathematician called Fourier 2 centuries ago!)
If we say a recording contains certain frequencies it means that the decomposition into sine and/or cosine waves contains sine and/or cosine waves at those frequencies.
For example a perfect square wave (that does not exist in the real world) is the sum of an infinite number of sine waves:
sin(x) + 1/3sin(3x) + 1/5sin(5x) + ...
If you listen to a square wave with a fundamental frequency of 10 kHz, the next harmonic will be at 30 kHz, the next at 90 50 kHz etc.
Inside the inner ear are hairs "tuned" to small frequency bands. For some hairs that small frequency band will include 10 kHz. They will resonate with the 10 kHz fundamental and as a result neurons will fire to signal to the brain that something happens at 10 kHz.
There are no hairs tuned to frequency bands that include 30 kHz, or 90 50 kHz, etc.
So you won't be able to hear the difference between a sine wave at 10 kHz and a square wave at 10 kHz.

The sine/cosine decomposition of a band limited signal (with 20 kHz upper limit) will only contain sine/cosine waves below 20 kHz.

The sampling theorem is a proven mathematical statement. What is proven is that if you have samples of the signal at least every half cycle time of the highest frequency in the signal then you can completely and precisely reconstruct that signal. All sine/cosine waves of the decomposotion can be reconstructed including the exact phase and timing! That is if the samples are exact. If the samples only have a limited precision, for example 16 bit, then the reconstruction will contain added noise. (However, 16 bits is enough to keep the noise far below audible levels, assuming you don't listen at such a high volume that the loudest possible parts of the recording will destroy your hearing.)

If you listen to a square wave with a fundamental frequency of 3 kHz there will be some hairs reacting to 3 kHz, some to 9 kHz, if you are lucky and not to old some to 15 kHz, and that's it.
If this signal is recorded and played back digitally with 44.1 kHz sampling rate then the 3, 9 and 15 kHz will be perfectly reconstructed, with the correct timing/phase.
If you listen to this recording you will not be able to distinguish it from the original square wave because all the audible components of the square wave are there (and the same hairs in your inner ear will be reacting).
(That is: unless your amp and speaker will cause audible distortion when fed with the original square wave, which is very likely because no amp can change it's output voltage infinitely fast, and no speaker membrane can move infinitely fast. In this case using the digital recording - or low pass filtering the original square wave - would lead to a audibly better reproduction of the square wave.)

[Edit: corrected a few wrong numbers...]

I didn't say that digital is inferior to analogue. Digital is potentially superior, but the question is how we can use it.
The sampling theorem is mathematically proven, but we need to have a better understanding of what is actually audible inside the analogue signal that we want to digitize.
We used to think that frequencies that cannot be heard separately cannot influence the perception of sound. However, what we hear in the first place in a waveshape, which is then analyzed into frequencies by the inner ear. Frequencies above 20 kHz that are part of the waveshape can influence the way lower frequencies are decoded by th inner ear from a timing point of view. In other words, the fact that frequencies above 20 kHz are not sampled can introduce phase distorsion in the perception of the audible range.
Even if we build a signal from frequencies that are stricly inferior to 20 kHz, the combination process will create higher frequencies inside the structure of the global signal.
These higher frequencies carry information about amplitude and phase, and this information has to be sampled in order to achieve high fidelity.

 

[audiokangaroo]

Sorry, no. We shouldn't admit that worst fidelity is better.

I share the opinion of 71 dB, and would add that it's not a topic on vinyl. Feel free to continue this in a proper thread, or make one if you care to discuss this, thank you.

The qulity of LPs can vary a lot. Some can be very disapointing because of the poor reproduction of the high range, but I have also experienced some that sounded much better than most CDs, in spite of the important amount of distorsion, which is the biggest of vinyl discs.

 

[bigshot]

Audiokangaroo, I would like to introduce you to the amazing world of the oversampling DAC. It allows for a low pass filter that doesn't cut into the top of the audible band. The best part is that just about every DAC out there today can do this. Hope this helps.

Inaudible isn't audible. Hope this helps too.

 

[71 dB]

Inaudible isn't audible. Hope this helps too.

At this point I am afraid this doesn't help, unfortunately. Digital audio theory together with the properties of human hearing just is a bit (pun intended) too complex of a totality for those who aren't willing to admit the holes in their understanding and knowledge. The way snake oil sellers muddy the waters with all the "inaudible things can be sensed somehow" nonsense doesn't help.

 

[bigshot]

So you think he actually believes all this? If that’s the case then he’s taking it on somebody else’s word. He doesn't seem to understand the things he's talking about.

 

[Davesrose]

Clearly audiokangaroo is not interested of learning anything, but instead tries to convince us about the "inferiority" of digital sound and strange ideas about what humans can hear. I made a few posts with patience, but at this point this patience is exhausted. I won't waste my time more. I am done with trying to explain to vinyl users that the reason why their CDs sound crap isn't because the format is inferior, but because crappy sound was for whatever reason put on the CD.

Why don't music producers add simulated vinyl distortions on CD masters on certain type of music genres sounding "better" on vinyl? People would LOVE the sound! On CD!

Reminds me of another poster who was saying all digital connectors are inferior. Even HDMI for video, and that you have to use component. Never mind the fact that component had up to 1080i resolution, and introduced many more stages of DAC>ADC>DAC>analogue video display.....consumer digital video movies give us 4K HDR (impressive that we're not just leap frogging with resolution, but now also dynamic range).

I admit that I have a record player and collected some older albums: for their masters....a good record that's cleaned can sound good. Those particular titles I got sounded better on vinyl vs their early CDs (where engineers might try to mix out tape hiss and not quite get dynamics). This is just a generality of course: and I do have classical CDs that are ADD or AAD that I enjoy. When it gets to music produced from 90s on (when all stages are digital), this is when it started to be a pinnacle of undistorted audio sources. It does amaze me that new vinyl records cost so much: I've only bought one because it has easter eggs and a hologram (maybe another advantage for some would be larger artwork). Most my day to day listening of music is with a HDD containing music files, or bluetooth with music streaming....feeding into a hybrid headphone amp (so if I want the analogue euphonics, I prefer tubes). I guess we'll have to always see folks claiming superiority of vinyl because it sounds good euphonically, and then try to justify that.

 

[theaudiologist1]

Wow I didn't expect this thread to blow up. I myself could not tell apart DSD64 and DSD128+ no matter how much people talked about the noise. The sound sciense forum really woke me up about pseudoscience and that 16/44.1 and DSD64 are more than enough. I'll still keep my hi-res files and vinyl rips, but I now know that they really aren't better than starndard CD, and least not in numbers.

 

[71 dB]

Wow I didn't expect this thread to blow up. I myself could not tell apart DSD64 and DSD128+ no matter how much people talked about the noise. The sound sciense forum really woke me up about pseudoscience and that 16/44.1 and DSD64 are more than enough. I'll still keep my hi-res files and vinyl rips, but I now know that they really aren't better than starndard CD, and least not in numbers.

Nice to see someone actually capable of learning something online, especially when it comes to the myths of digital hi-res audio.

 

[gregorio]

[1] Digital is potentially superior, but the question is how we can use it.
[2] The sampling theorem is mathematically proven, but we need to have a better understanding of what is actually audible inside the analogue signal that we want to digitize.
[3] We used to think that frequencies that cannot be heard separately cannot influence the perception of sound. [3a] Frequencies above 20 kHz that are part of the waveshape can influence the way lower frequencies are decoded by th inner ear from a timing point of view. In other words, the fact that frequencies above 20 kHz are not sampled can introduce phase distorsion in the perception of the audible range.
[4] Even if we build a signal from frequencies that are stricly inferior to 20 kHz, the combination process will create higher frequencies inside the structure of the global signal.
These higher frequencies carry information about amplitude and phase, and this information has to be sampled in order to achieve high fidelity.

1. No it's not. Your argument, that vinyl is "globally better than CD", is a question of the relative merits of the distribution formats, NOT the content on those formats, "how we can use it".

2. Science has been studying this issue for over a century and for many years we've had a perceptual model that's effectively perfect. Why do "we need to have a better understanding" than one that's already effectively perfect?

3. Yes, we did used to think that and we still do because there's no reliable evidence to suggest otherwise!
3a. No, during many decades of research there's no reliable evidence that supports that assertion but plenty that discredits it!

4. No, audio freqs higher than 22kHz carry no more amplitude or phase information than freqs below 22kHz. Audibly perfect fidelity is possible even with 44.1/16, however, audibly high fidelity is the limit of vinyl. That's why your argument appears to make no sense, effectively that perfect fidelity is worse than high fidelity. Although, it's a bit difficult to work out your argument because if having these freqs above the 22kHz CD limit are necessary to achieve high fidelity and as vinyl doesn't have any accurate reproduction of >22kHz content then presumably vinyl isn't high fidelity either?

G

 

[audiokangaroo]

1. No it's not. Your argument, that vinyl is "globally better than CD", is a question of the relative merits of the distribution formats, NOT the content on those formats, "how we can use it".

2. Science has been studying this issue for over a century and for many years we've had a perceptual model that's effectively perfect. Why do "we need to have a better understanding" than one that's already effectively perfect?

3. Yes, we did used to think that and we still do because there's no reliable evidence to suggest otherwise!
3a. No, during many decades of research there's no reliable evidence that supports that assertion but plenty that discredits it!

4. No, audio freqs higher than 22kHz carry no more amplitude or phase information than freqs below 22kHz. Audibly perfect fidelity is possible even with 44.1/16, however, audibly high fidelity is the limit of vinyl. That's why your argument appears to make no sense, effectively that perfect fidelity is worse than high fidelity. Although, it's a bit difficult to work out your argument because if having these freqs above the 22kHz CD limit are necessary to achieve high fidelity and as vinyl doesn't have any accurate reproduction of >22kHz content then presumably vinyl isn't high fidelity either?

G

1. CD has a cleaner sound but it doesn't sound natural. Vinyl has more noise but it sounds more like actual instruments.

2. Science has still along way to go to explain digital sound problematic. The sampling theorem is not sufficient. Applying this theorem does not provide a high level of
audio fidelity.

44.1/16 PCM is far from achieving high fidelity. If you compare it to DSD256, you can hear that DSD sounds much more natural and true. This should make us admit that capturing frequencies above 22 KHz is necessary to achieve high fidelity audio, as DSD256 is at least equivalent to 2000 KHz PCM, in spite of a lower signal to noise ratio in high frequencies.
However, I'm not sure that non acoustic genres, based on electric guitars or synthesizers, require such a level of fidelity

 

[vergesslich2]

Off-topic, regarding high bandwidth:

Shouldn't the "wave field" be as interesting as ultrasound? I just wondered why the topic was more about ultrasound, kind of. With a multi-channel setup (2, 3 and up) one would make use of more digital bandwidth, and one could reason about interference of some kind? At least I thought so. ... The form factor is just not as practical. I tend to buy a dongle with nonsense capabilities rather than a subwoofer. Maybe that has something to do with it.

Another point:

There's a Youtube video of someone saying he thinks vinyl is more realistic, without saying why exactly. I wondered if his definition of realism could be that degradation of measurable qualities helps him. E.g. when the playback device is randomly taking out some frequencies or changing playback speed.

That could somehow be like the situation where you want only one person to talk instead of everybody, or when you solo a channel in your mix. If there was variance of what is "solo'd" over time, since vinyl and cassette might change their sound somewhat constantly, the overall perception result could be that of increased realism, because a listener might increase introspection into the music, when they play it often.

I do think snake oil is sold and is wished for, for obscure reasons. But regarding the "downgrade oil" I must admit that I was recently thinking about buying a portable cassette player for fun. Not really to increase my insight (well, maybe) into music, but to be inspired. That is what this thread reminds me of.

 

[old tech]

Off-topic, regarding high bandwidth:

Shouldn't the "wave field" be as interesting as ultrasound? I just wondered why the topic was more about ultrasound, kind of. With a multi-channel setup (2, 3 and up) one would make use of more digital bandwidth, and one could reason about interference of some kind? At least I thought so. ... The form factor is just not as practical. I tend to buy a dongle with nonsense capabilities rather than a subwoofer. Maybe that has something to do with it.

Another point:

There's a Youtube video of someone saying he thinks vinyl is more realistic, without saying why exactly. I wondered if his definition of realism could be that degradation of measurable qualities helps him. E.g. when the playback device is randomly taking out some frequencies or changing playback speed.

That could somehow be like the situation where you want only one person to talk instead of everybody, or when you solo a channel in your mix. If there was variance of what is "solo'd" over time, since vinyl and cassette might change their sound somewhat constantly, the overall perception result could be that of increased realism, because a listener might increase introspection into the music, when they play it often.

I do think snake oil is sold and is wished for, for obscure reasons. But regarding the "downgrade oil" I must admit that I was recently thinking about buying a portable cassette player for fun. Not really to increase my insight (well, maybe) into music, but to be inspired. That is what this thread reminds me of.

It could be the reason, particularly for someone who grew up with the vinyl sound, coupled with ageing ears, that it may sound more realistic to them. Some people also prefer a more distorted sound over true fidelity. I'm a bit partial to the bass sound from analogue tape when recorded at tape saturation levels. Though of course, that sound can be reproduced perfectly by digital media.