• WANTED: Happy members who like to discuss audio and other topics related to our interest. Desire to learn and share knowledge of science required. There are many reviews of audio hardware and expert members to help answer your questions. Click here to have your audio equipment measured for free!

Kali Audio IN-8v2 (Second Wave) 3-Way Studio Monitor Review

Do this experiment. Use a transient shaper plug-in. Set the attack fast once, slow once. Does the sound sound the same? No. It's the same with speakers, some have a fast attack, others slower. You can add high frequencies and the attack won't speed up, because the impulse matters. Now verify your knowledge and open your mind to the unknown. You'll thank me later.
Transient shapers just use dynamic expander and compressor to shape the attacks and shortens or lengthens the sustains. Which basically is amplitude modulations/manipulations (modifying the waveform amplitude envelope). Picture source here.
51a3300a9ba7-drum-break-with-fast-attack-and-short-sustain.png.webp


Here is how the effect will show up in a typical Fourier analysis. Both have the same fundamental frequency. The "fast" and "slow" waveforms are normalized to have the same total energy. The fast has faster attack and quicker decay. First plot shows the waveforms, second shows the spectra. The fast one doesn't require significantly higher bandwidth to reproduce.

Since these are not a steady signal, pure frequency response curves are not the most sensitive indicators of differences. There are other more suitable ones, e.g. spectrograms or waterfalls.

waveforms.png

spectra.png
 
I recently purchased these speakers for home listening. They won in comparison with Adam t8v at medium frequencies, although they lost in the scene and high frequencies. There are minor flaws in both Kali speakers purchased: one speaker has a strange bzzz modulation at low volume, the frequency of which depends on the volume, the support said that the matter is resonant inside, although, it seems to me, the sound is strongly electronic, the other has a slight cosmetic defect. It does not interfere with listening, but be careful if you are a perfectionist.

I have a few questions:
1. Do the switches on the back panel only change the frequency response? Is this the same as me using an equalizer? Or is there something trickier and, regardless of the use of the equalizer, it is worth putting the switches in the correct position? I noticed that the best vertical listening point differs depending on the position of the switches, but I don't know if it's solely about the frequency response or, for example, delays between the speakers.
2. Do I understand correctly that there is no point in thinking about the sound source and my ESI Juli@ sound card is enough? If it is wrong, what level of source is better to use? Maybe with a headphone amplifier, otherwise I don't like my daart canary and I'm thinking of changing it. Does it make sense to take a device with a configurable dsp when listening from a computer? Using an equalizer helps at frequencies below 200 Hz, although Charles Sprinkle wrote: "EQ applied to a bad setup is like putting lipstick on a pig. It's still a pig. " :)

And theoretical, slightly flood questions:

There is a feeling that some apparent sound sources are tied to the treble speakers, they do not sound in the general scene. How do I check if this should be the case, given that this is the best system I've ever had? :)

We have a DAC and chipboard inside, but we have to think about buying a DAC and using a software equalizer, why not give the user digital interfaces?

Why don't they put a piece of paper with measurements, since they measure it anyway before release?

If speakers are recommended and used for audiophiles and there are fans of phase linearity and zero group delay among listeners, for example, in a neighboring topic, a person scolded the phase, but did not present measurements, why not make a dip switch linear phase system - real-time
system?
 
There are minor flaws in both Kali speakers purchased: one speaker has a strange bzzz modulation at low volume, the frequency of which depends on the volume, the support said that the matter is resonant inside, although, it seems to me, the sound is strongly electronic, the other has a slight cosmetic defect. It does not interfere with listening, but be careful if you are a perfectionist.
Kali has in fact had buzzing issues before; that time it turned out to be the midrange wires that had not been covered in foam. I assume this should not be too hard a fix for those mechanically inclined. (My old Tascam monitors were notorious for relay issues, and those had been screwed and glued shut. That sure got them airtight but made them hard to open without some ingenuity and brute force. I never bothered.)
1. Do the switches on the back panel only change the frequency response? Is this the same as me using an equalizer?
Yes and yes (the parametric kind, obviously).
From Erin's review:
Kali%20IN-8v2%20DSP%20Boundary%20Settings.png

2. Do I understand correctly that there is no point in thinking about the sound source and my ESI Juli@ sound card is enough?
1. Since you are presumably using the card with the balanced output side, be mindful of the fact that it can dish out up to +20 dBu, which may be more than the Kalis' input can handle. You may want to turn one of them waaaay down and output a maybe 1 kHz sine at various levels to see if or when the input clips. I mean, the specs suggest that they should be able to take at least +24 dBu (97 dB SPL @ +4 dBu, 117 dB SPL peak), but... Maybe avoid turning them down further than needed to minimize hiss.
2. The Juli@'s ADC arguably has stood the test of time better than its DAC, which is a bit meh. While dynamic range is fine, the digital filter has so-so levels of periodic ripple (+/-0.02 dB @ about 3 kHz), and I would advise using good-quality upsampling to 176.4 or 192 kHz to minimize its impact. (BTW, is it known what converters the Kalis are using?)
3. You're probably using Linux then? Apparently the card works fine there... the Windows drivers have some inexcusable flaws. No clue about available upsampling options though.
 
Thanks for the answers!

The sound bzzz appears at a very low volume, I only heard it at night when I was looking at the limits of audibility. It is best listened to on the bass when the bass itself is not audible and only one speaker has it. As the volume increases, the sound becomes higher in frequency and either goes beyond the line of hearing, or is masked by a useful sound. It does not interfere with real use. It seems to me that even if some large part of the monitors has such an effect, people simply do not notice. I came across it by accident during a specific test. I can send an entry if you are interested.
1. from the specifications of Juli@: Unbalanced RCA with -10dBv or Balanced TRS with +4dBu https://www.esi-audio.com/products/julia/
In a room of 20 square meters in an apartment building, it's too loud. I tried, but I didn't maximize the signal. The microphone picks up noticeable distortions. I'm not sure if this test is really necessary, because I don't listen so loudly, although I can probably somehow connect from my tablet via TeamViewer and lock myself in the bathroom or leave the apartment :) The interesting thing is that at high volume on the bass, the speakers are ventilated and there is a noticeable smell of plastic :)
2, 3. It is possible to set the sampling rate in the sound card control panel. I use Windows 10 and Equalizer APO. Of the disadvantages, only the rare appearance of noise when changing the volume in the system. To get rid of it, you have to change the volume several more times. I've heard about the problems with using vst plugins in professional activity, but I don't use them. The latest driver was released on 2023-05-29.
панель юльки.png

In a good way, it is better to invest resources in room sound treatments and not change the sound card/DAC yet. But thoughts about whether it's worth buying something better still visit.

BTW, is it known what converters the Kalis are using?
I think
ADC PCM1863
DSP STM32/ARM Cortex M3
DAC and power amp
TI TAS5805M
TI TAS5825P

The Kali PowerPoint presentation was admirably forthcoming about the silicon side of their "2nd Wave" stuff -- these chips are very sophisticated, highly integrated devices, folks.

TI TAS5805M

TI TAS5825P
For Kali IN-5:
there's an STM32/ARM Cortex M-3 chip for that fed by PCM1863 ADC
 
Last edited:
The sound bzzz appears at a very low volume, I only heard it at night when I was looking at the limits of audibility. It is best listened to on the bass when the bass itself is not audible and only one speaker has it. As the volume increases, the sound becomes higher in frequency and either goes beyond the line of hearing, or is masked by a useful sound.
That sounds like it's emanating from an inductor in the power supply then (they tend to go into burst mode at low load). It's audible even when nothing plays, right? This may be bad luck in terms of components then, maybe a loose winding.
1. from the specifications of Juli@: Unbalanced RCA with -10dBv or Balanced TRS with +4dBu https://www.esi-audio.com/products/julia/https://www.esi-audio.com/products/julia/
I know, I've read the specs in the manual. Maximum +6 dBV unbalanced and +20 dBu balanced.

So which card configuration are you using?
In a room of 20 square meters in an apartment building, it's too loud. I tried, but I didn't maximize the signal.
LOL, I wasn't intending you to try and tear the house down... the idea was to turn down the knob labeled "Volume" on the back first. You could hit peak levels of 113 dB (1 m, anechoic) at the default setting of 0 dB, which I rather assume is like 6-10 dB more than you'll ever need... and maybe another 10 dB if you're a quiet listener.
2, 3. It is possible to set the sampling rate in the sound card control panel. I use Windows 10 and Equalizer APO.
Then I'd just leave it at 24/192 and output everything in shared mode. While I'm using the superb and computing time friendly SoX resampler DSP in Foobar2000, Windows 10's upsampling is perfectly fine as well, just be mindful of the -0.1 dBFS hard-limiter in the audio stack. Using ReplayGain or other similar playback volume normalization is recommended, and if your musical diet includes very dynamic classical orchestra works you may find a need for several dB of negative preamp for RG assuming the common -18 LUFS target in Foobar2000 etc. (I've had -3.3 dB set for years now and will accept that more extreme cases get turned down a few dB by the "avoid clipping" option, otherwise I'd need almost -8 dB - good ol' Mahler). You've got a bunch of dynamic range available now, might as well be using it.
The latest driver was released on 2023-05-29.
Oh wow, they released drivers for the "old" (PCI) Julia@ as recently as this? I gave up and mothballed mine probably a decade ago (they sure look nice in the packaging). At that time the drivers hated standby and hibernation with a passion, which was completely unacceptable to me. Used an Asus Xonar D1 for a long time (also quirky but workably so), still have two of those and a D2 floating around. My current PC no longer even has PCI slots, and after playing around with a Xonar SE I decided the extra hassle wasn't worth the increased power consumption (the thing doesn't like PCIe ASPM) and went with the onboard Realtek ALC1200 instead. Technically a downgrade but good enough.
Got the machine down to ~14.5 W in idle (not including monitor and speakers, which are using about 3 times as much combined). Load is a different story, of course, Rocket Lake was the last hurrah for 14 nm and is not known for its efficiency... (I can put on the brakes with ThrottleStop though.)

I think
ADC PCM1863
I'd hope Kali used the (better) IIR filter option then.
DAC and power amp
TI TAS5805M
TI TAS5825P
Hmm. Not a word on digital filter performance on the datasheet. I suppose it wasn't one of their bigger worries.
 
Last edited:
It's audible even when nothing plays, right?
No, only when there is a signal
So which card configuration are you using?
I use a balanced output
the idea was to turn down the knob labeled "Volume" on the back first.
Ah! I thought the volume control was at the input, just to match the levels. But you're right, it changes the noise level, so it's not just standing at the entrance. The volume control decreases by about 18dB, then a cutoff is triggered and the sound becomes zero. It's also not quiet at the maximum volume sine :) Does it seem possible to say that the input is not overloaded? Z-weighting(flat), RTA240 bands
регулятор громкости.png

just be mindful of the -0.1 dBFS hard-limiter in the audio stack
Thanks! I didn't know about it!
You've got a bunch of dynamic range available now, might as well be using it.
I set -20dB ReplayGain and used it for the equalizer :) If I understood correctly and there is no input overload, is it enough just not to cross -0.1 db?
At that time the drivers hated standby and hibernation with a passion, which was completely unacceptable to me.
Precisely! There is a very annoying problem with clicking when turning on the computer. I have to turn off the monitors. How could I forget to write about it!
Realtek ALC1200 instead. Technically a downgrade but good enough
I know there are good built-in sound cards on laptops now, probably realtek too. I like them less than Julia, but they are, to my taste, about the same level as Daart Canary. Without relying on objective measurements and audio science
 
I was thinking of trying a mid- and high-frequency equalizer as a first approximation. It was interesting whether it was needed or not. I uploaded the data of Erin and Amirm to REW, it turns out that they are significantly different. Probably, these are the differences between versions 1 and 2. I would not like such differences to be between instances.
1727467164027.png
 
Last edited:
Back
Top Bottom