I think I suggested you try this sort of approach with HQplayer, must have been close to a year ago and explained it's an easy way to short cut design decisions. All of the modulators, digital filters, OP formats are right there to use and help you prioritize a forward path for DAC mods.
I guess now it's a great idea.
T
Many years ago I gave a friend, a very smart EE, a few nice opamps (OPA627) to replace the TL072's in his Marshall Silver Jubilee. They were used in effects loop for buffer / gain make up etc.
He said I was wasting my time and they could never possibly make any difference because of all the preamp tube distortion.
When I next saw him, he said your not getting those opamps back, what do I owe you?
T
I guess now it's a great idea.
It wasn't applicable for what I was working on then. It might be something to try to help investigate something different. May or may not be helpful for looking into the reverb tail issue, but it would be an easy thing to try. You can take credit for telling me about it in case a need ever came up.
Speaking of a need coming up, I will probably try HQplayer for AK4499, but that will be a new project. FYI, the preliminary data sheet for AK4499 eval board says it includes an AK4137, the chip I did use a year ago for the project I was working on then.
Ok, regarding the current discussion regarding 'memory', reverb tails, Hirata waveform etc.......
I perfectly reliably encounter a curious effect when rapidly switching between same time point in 'same but different' files causing ABAB to sound different to AABB into speakers or headphones for the first few seconds. It is as though the first B file play overwrites some kind of 'history' or embedded behaviour implanted by the A file plays. The inverse condition also applies, ie when switching from B file back to A file this 'implanting' sequence occurs anew.
I just regard this as an 'initial run-in' behaviour that takes the A file conditioned system behaviours to a new 'setting' at the first high/highest amplitude peak in the first B file play, with little to none change after the B file first high/max amplitude peak transient event.
On rock/pop music the required peak events generally occur soon into the track, but orchestral/natural etc that starts low in level may take some time before delivering a high amplitude 'set' transient that 'locks in' new/full B file behaviour. When switching files on the fly (multitrack editor/DAW), there is delay before the new file system sound stabilises. IOW when switching from A file or item conditioned system, to B file or item there will be a variable delay according to the programme before the system is fully in B 'mode'. IMHO to those unaware of this signal embedded noise dependent system behaviour embedding this is a major confounder if not THE major confounder in ABX testing and reinforces longer term listening findings.
I have thought of remedies that I have not rigorously trialed yet, and that includes a 0dB or so, one or two cycle 10kHz (maybe 20Hz ?) transient or something like that to inaudibly 'exercise' the system to peak or near peak values at the beginning of each music track. This conditioning pulse/waveform could be locally generated and constitute a 'system reset' between tracks, or could be embedded into the start of each file where it would serve as a 'system set' and ought to reduce one aspect of signal/time dependent system behaviour.
I offer again a bunch of 'same but different' wav files that will help to illustrate this behaviour and enable awareness, perhaps result in measurements protocols. Bennie Goodman 1938 Jump Lana Del Ray Test Dept - Two Flames So grab these recordings, see what differences you discover, ask, discuss, suggest other recordings.
Are the infamous dirt, potato etc recordings (Pano ?) still available ?....I am interested to take a closer listen. How applicable is the Hirata waveform in Wav file form.....anybody have a copy or able to generate Hirata wave files ?. Scott, is there a ready/easy method to perform your octave/half octave etc filtering to existing music files......Audacity/Audition/Reaper etc processing preset ?
Dan.
It can, but if you have high frequency hearing loss is could affect your perception of the attack
Is the AK4490 know to have this sort of dynamic instability? I'm currently trying to measure effects like this looking down 30dB++ into the noise but to no avail as of yet. How long is the "state variable recovery time"? Is the instabiltiy deterministic, always the same when a loop of an audio snippet is played endlessly?
Did we ever get a definition of what a reverb tail is? I may well have missed it.
Not really the same - the equivalent would be to set the first fuzztone to a distortion level 80dB below the second one, then tray and hear it.
I would expect at least the DSP version to not suffer from what I surmise to be a slight time alignment dip around the xover frequency. Both the KEF version as the modified xover show it, albeit shifted in location. On the whole, dispersion is determined by the physical characteristics of driver and enclosure. It is not possible to correct for unwanted behaviour in this domain by tweeking the electrical side. In other words, if you were to correct the sound power output of this loudspeaker so that you would obtain the desired down slope, the on axis FR would suffer (dip).
Perhaps but I'm not sure how or if it has a significant effect - can you say more?
Yea, I'm not sure either - I had initially assumed Scott was talking about the release portion of a sound, the part that comes after the sustain portion?
Like what?
You could experiment on yourself with a sharp attacking plectrum plucked bass note for example and a low pass filter.
While you are being funny, did you take a listen to the files I linked Indra1 ?. How about you scottjoplin ?.....the Benny Goodman live recording was on your recommendation (even though different version). You guys should be able to hear some difference between the 01 and 02 files in each folder, and if not the certain conclusion is that you have inadequate systems and/or profound deafness and there is no point to proceed further....you can continue to blissfully enjoy your music sub-optimally but you must understand that you are in no position to make declarations about other's systems or other's hearing acuity.You need to ask someone like Dan for something like that.
If you do report 01/02 file differences accurately you can progress to determining 01/03 file differences and then onto 02/03 and 02/04 file differences. If you can correctly describe 02/03 and 02/04 file differences you are at high listening skill level and can proceed to 03/04 differences. If you can describe 03/04 file differences accurately you are at master class level. If you can achieve master class level I am happy to discuss the hows and whys of what I am doing, it's up to you armchair experts and critics to stump up and show your real skills and understanding of the art. Dan.
Last edited:
I love my KEF’s. Precious few speakers image as well or are as clean and open sounding.
This discussion is like the Oppo’s. Everyone loved them and they represented great value for money and then a few decided they were no longer cool.
Like what?
You seemed to indicate you have experience of this & can say what the effect is?
Since we're speculating without data, I'll vote for: I doubt age-related high-frequency loss will change the perception of the attack of a low frequency sound. There are 2 main reasons for age-related hearing changes, and insertion of a low-pass filter in the ear is not one of themLike what?
The dominant reason is loud-noise induced loss of hair cells (the mechanical-to-neural transducer in the inner ear) that are sensitive to high frequencies. That does not change the attack appearing to the HCs sensitive to lower frequencies. The second reason, loss of both elasticity and stiffness in all connective tissues in the body (look in the mirror, if you're old like me), could, perhaps, but I'd have to look into that more... I suspect not.
Yes, Mark is talkin' about the release/decay portion of sounds......where the initial sound tails and reverb tails descend into the noise floor.
Seemingly minor system changes can make quite profound changes to sounds as they fade into and below the noise floor....these fades/decays are complex and this causes them to be revealing of 'minor' system changes.
Dan.
Last edited:
- Status
- Not open for further replies.