Preamp difference : if it's not the frequency, not the slew rate, and not the harmonics, what is it ?

GroupDIY Audio Forum

Help Support GroupDIY Audio Forum:

This site may earn a commission from merchant affiliate links, including eBay, Amazon, and others.
My studio monitors, made by Wayne Jones Audio, now come with carbon fiber boxes and the new model does sound different than my model without the carbon fiber. Not a huge difference, but noticeable.
of course the sound not leaking out the back, comes out the front. 🤔

Cabinet rigidity was an issue mostly for bass response in molded plastic cabinets (back last century when I was over all product management for Peavey). Adding a wood internal strut to stiffen up the plastic cabinet helped, but the extra weight worked against the light weight selling feature.

JR
 
of course the sound not leaking out the back, comes out the front. 🤔

Cabinet rigidity was an issue mostly for bass response in molded plastic cabinets (back last century when I was over all product management for Peavey). Adding a wood internal strut to stiffen up the plastic cabinet helped, but the extra weight worked against the light weight selling feature.

JR
These speakers, even without the carbon fiber, have heavy, rigid solid wood cabinets. The monitors weigh 45 pounds, so I assume any rearward sound is diffraction/baffle step below 250 Hz or so. The builder says the carbon fiber reduces cabinet resonances, which would be internal and would not affect leakage.

Plastic speaker cabinets sometimes benefit from a layer of dynamat or mass loaded vinyl, too.
 
The speaker enclosure material should not be resonant, have great damping, have strength, easy to work with, looks not relevant to sound output. Added mass is a bonus. Concrete is probably overkill. Granite should work fine, not the easiest to cut. Fred Flintstone would approve. I'll stick to MDF.
 
Hi,

I took the opportunity to measure different preamps (new SSL ones, Focusrite Scarlett, Warm Audio TB12), and investigate a bit on the topic of "audible differences between preamps".

If we run the preamps too hot, I came to the following conclusions :
  • The preamps are usually rather flat in terms of frequency. At least nothing that couldn't be fixed with some basic digital EQing. Example : Measurement of an API312 here on SoundOnSound
    So it seems unlikely that frequency response is really what differs between preamps.
  • The harmonics are of course different, in the case of transformer-balanced preamps. But they are usually at a rather subtle level (unless the preamp is pushed).
  • The slew rate of the preamp usually seems to allow for a correct reproduction of all the audible frequencies.
So now I am wondering :
  1. Why would a preamp like the API be called "punchy" ? What would "punch" be ?
  2. If it's a variation of the transients (we hear about "slow" vs "fast" preamps), then where does it come from if the slew limit is > 20000 Hz ? The transformer ? If so, aren't the transformers supposed to be rather transparent, harmonics put aside for the lower frequencies, in the frequency range ?
  3. Would the slew rate, if a bit too low, be able to influence the audible frequencies (even if the amplifier would, on paper, be able to reproduce those frequencies without any problem) ? It seems so : https://hifisonix.com/wp-content/uploads/2018/03/SID_and_TIM_W_Jung_77-79.pdf
  4. Did I miss any phenomenon ?
I've searched all around the Internet but I can't find a clear answer.
Everyone seems to hear punch and smoothness differences, and I feel I hear them too, but where does it actually come from ? Has anyone been able to 'demonstrate' that ?

Thanks
Adrien
Hi Adrien
I made the same observation and during my various projects I tried to understand what could be happening. My (provisional) conclusion is that these differences are attributable to dynamic and not static aspects because measuring harmonic distortion in steady state on a sinusoidal signal is (I think) not very representative. When the components are well chosen and well used, the effects of DHT, slew rate and noise at these infinitesimal levels do not seem to me to be detectable by the human ear. The level differences and therefore the "punchy" aspect that you speak of are not integrated into this measurement protocol. The load impedance in relation to the current capacity must have a probable influence on this parameter. Finally, for line stages or microphone stages integrating an audio transformer, the characteristics of this component are essential for the sound signature. If we talk about legendary analog consoles like the Neve or SSL, they are all made from NE5532 or 5534 type circuits and even some with TL07n. However, these circuits are very inferior to what we can have today and yet the sound of these consoles is very appreciated and often considered "better". Is this a collective illusion linked to psychoacoustics. I don't have the definitive answer but I'm not entirely convinced that this is the case. What I noticed is that nowadays OPAMPS are often designed with consumption constraints which perhaps limit dynamic performance. In addition, I deliberately ignore the fact that the open loop bandwidth is very limited and the gain is very high which thanks to the feedback allows a high bandwidth and very low distortion but be careful, this is measured in steady state and not in impulse mode. However, under these conditions, the internal amplifier stages can go into saturation for a very short time and cause audible differences.Daniel
 
Sorry for the veer, but which DAWs did you test? I've long heard differences in DAWs, which many claim don't exist, but never really quantified it. One thing I've noticed is many don't dither properly.
I haven't read through this entire thread yet. Was just searching "dither" for a question I have and came across this. Will read later today when I get back.

My question, and apologies if it's already been mentioned, is...

If playing a 24bit file from the DAW, no internal processing (volume changes,etc) through the DAC, through an external effects device, and back through an ADC recording at 24bits, there wouldn't be a need to apply dither?
Only if there were any processing done in the DAW would there need to have 24bit dither applied before it headed out to the DAC?
I hope this makes sense?
 
I haven't read through this entire thread yet. Was just searching "dither" for a question I have and came across this. Will read later today when I get back.

My question, and apologies if it's already been mentioned, is...

If playing a 24bit file from the DAW, no internal processing (volume changes,etc) through the DAC, through an external effects device, and back through an ADC recording at 24bits, there wouldn't be a need to apply dither?
Only if there were any processing done in the DAW would there need to have 24bit dither applied before it headed out to the DAC?
I hope this makes sense?
No dither is required if not reducing the bit depth - it’s normally only used when going from 32 or 24 down to 16 bit to reduce quantisation errors by introducing random noise. If you’re not changing the bit depth, don’t use dither.
 
No dither is required if not reducing the bit depth - it’s normally only used when going from 32 or 24
Yes I understand. Just wanted to make sure . To be clear, if I were to make a simple gain change or any other type of moves or processing in the DAW, I should have 24bit dither activated on the master out if wanting to record back in 24bit?

Is it common practice to have dither on at all times when working on a mix project where obvious higher than 24bit processing will be happening in a project? Basically monitoring with the dither on since interfaces and converters are usually 24bits?
 

Latest posts

Back
Top