Moving to new location

This blog continues on the Kvålsvoll Design site:

https://www.kvalsvoll.com/blog/en/

Acoustic absorber building description

Acoustic absorber building description

Several acoustic absorbers with different shape and size were made for the new Room 2 listening room. 

Building description with construction drawings:
www.kvalsvoll.com/Designs/absorber120x120x20/absorber120x120x20_buildingdescription_en.pdf


Presentation:
www.kvalsvoll.com/Designs/absorber120x120x20.htm

Why Vinyl Sounds Better

Vinyl sounds better, right? If so, there is a reason that can be explained in technical terms. Let us find out. 

The answer is simply that vinyl does not sound better, in fact it is inferior to digital.

Why vinyl sounds better

Today was a good time to find out why vinyl (what we called turntable and records in the old times) sounds better. 
So the plan is to rip some records I have, some of which are not available on digital.
There are some good stuff here that may make its way to the demo list as well. 

The preamplifier 

Central to this exercise is this preamplifier, it is the last one I made in late 80's. 

It has MC input only, for pickups like MC30 and AT OC9.
It had a shunt-regulator power supply burning continuous power as a medium power amplifier long before this became a trend in the audio world.
 
The problem with preamps for MC is noise and low-level distortion, because the signal is so low that the thermal noise of the source generator impedance becomes significant.
The best one can achieve is to get as close to this physically given constraint as possible.
FET transistors that have long been out of production and now impossible to find solves this problem, providing extreme clarity and resolution but at the same time comfortable, easy-to-listen-to sound.


The preamplifier for moving coil pick-up

I was never satisfied with any of the front plates I made, so it usually ended up without.
The volume knob is massive brass, it weighs so much it destroys the potentiometer, and that is how it is supposed to be.

Is it better 

First part is to find a suitable vinyl record which I also have in digital format, where it is likely that the master is similar. So that the mythical vinyl sound can be compared to the digital version, and hear what is better on vinyl.

Well, it turns out that it really does not sound much better. 
The differences can be attributed to differences in frequency response, with eq in one or the other it sounds pretty similar.
Part from a few things that are significantly worse on vinyl. 
The noise level is so high that the decay and room information are somewhat masked, the scene is a bit smaller and narrower, the bass is more powerful and less precise, higher frequencies less sharp and more flat.

No, it is definitely not necessary to do a blind test to verify that there is a difference..

On some albums the vinyl may sound better overall because the noise masks quite a lot, and added distortion can give more "musicality".

Technical analysis

Comparing vinyl to digital reveals there is no significant difference in dynamics:

Vinyl is the first graph.

The allpassed crest is flat for both, and the small difference can be attributed to differences in spectral balance, the vinyl is tilted towards more bass and less highs:

As usual, this corresponds well with listening impressions.

But what with the claims for higher measured dynamics for vinyl?
In some cases the reason is different masters, and additional difference in measured crest is due to use of limiters, which causes crest to increase when played back on vinyl due to phase changes.

Advantages from ripping vinyl to digital

Using a reference-quality AD-converter and computer it is possible to do digital recordings of the vinyl records, with no loss of sound quality, the resulting digital files will sound exactly like the sound coming from the exotic preamplifier.

Convenience - can be played back just like all the other music, is one obvious advantage.

But the sound will also be better.

It is now possible to fix faults introduced in the vinyl playback process, with digital signal processing on the sound files.
Improper channel balance, faults in frequency response, can be fixed.

Pop and cracks can be removed - a very audible improvement.

In my system there is digital signal processing for crossover and bass management, requiring any analog input to be converted to digital. Potentially, this additional conversion can affect sound quality - may be it doesn't, but now that the source is digital like everything else, this problem is eliminated.

And yes, the ripped digital files sounds exactly like the original vinyl. And will continue to do so, without loss of quality due to vinyl wear.

Vinyl sound is just another audio myth

So, this was not a step in the right direction towards audio nirvana.

A record player also means a lot of nuisance.
It has to be leveled, adjustments and alignments for the pick-up, dust removal.

Yesterday I listened to an Opus3 recording, the one with the track "Tiden bara går". 
Fantastic, with Therese Juel visiting in my room, exactly as I remember it from the old days.
I downloaded an mp3 of this track recently (it is provided as a free sample) and it does not sound like this at all, this is one reason I was tempted to set up the turntable again, to see if that magic from the old day was still there.

But when comparing even good quality vinyl records to decent digital, digital is preferable. 

I will rip a few records, then the exotic preamplifier and the record player is going back to where they belong - as a show-off in a storage room.

About the author

I have now presented short technical articles on sound here, for some time. Who am I, what is my background, what is my intention.

 

About the author

Intention

I want to increase interest for sound and sound quality.
I like good sound, and want to contribute to increased awareness and focus on sound and what it adds to the experience of music and movies.

Presenting facts and informative articles is one of the ways I have chosen to contribute.

Current status of the blog

This blog presents articles with a very specialized and narrow perspective - effort, knowledge and interest in sound is required to understand and benefit from the content.

So this is for the experts, for those with a special interest for sound related technology.
That narrows the potential audience down to a small group of people.

After running this blog for two years, the audience is still very small, and that is the main reason for the rather low frequency of updates, there is not much point in writing if there are no readers.

I recommend my company web pages for more articles and information on what I am developing. 

About the author

This text is taken from the article How to set up a Home Theater Sound System on the Kvålsvoll Design company web page:


I run the Norwegian registered company Kvålsvoll Design AS.

I like to create things, I like to work on engineering challenges as well as shape and visual appearance. I also enjoy working on abstractions like systems and concepts. This forms the foundation of Kvålsvoll Design.

I am an engineer in cybernetics and electronics. I have started several companies, and I have designed, built and tested complex cybernetic systems involving advanced dynamic control algorithms.

I designed and built my first speakers around age 14, and continued exploring and building audio speakers and amplifiers as a hobby throughout the years, learning more and applying what I learned in school about electronics, system theory and mathematics.

In 2012-2013 I started designing loudspeaker systems for home entertainment. Development is based on my expertise on simulation and dynamic systems. I have also learned a lot about how we perceive sound, which is of vital importance for defining proper requirements for sound system design.

The development of audio solutions in Kvålsvoll Design has lead to several new technologies for sound reproduction, such as the Compact Horn subwoofers, controlled directivity waveguide horns, very-high-slope crossover networks.

Acknowledging the importance of set-up and calibration, a new method for subwoofer calibration has been developed, which accounts for sound-field intensity for better tactile feel. This is new, and Kvålsvoll Design is so far the only company using this method.



Thank you for reading,

Øyvind Kvålsvoll

Audio in 2015 - a summary of trends

Norsk versjon / Norwegian version ->

This is meant as an update on audio for those who have been out of the game for a while.

Some technology improvements, but most of all the wide-spread communication on internet, has made it possible to get huge improvements in sound quality.

The new sound

1. Dynamics and realism with high efficiency speakers

It's the new sound trend - dynamics, realism, more life-like.
Speakers with higher efficiency, more output capacity and controlled sound radiation pattern is the key.
Often larger, with horns, often utilise professional loudspeaker drivers.

Dome tweeters and small satellites with small woofers are out - large woofers, horns and waveguides are in.

Big is back and small is now old-school

2. Acoustics

We always knew the room and furniture in it, along with placement of loudspeakers and listening position, have a significant effect on sound quality.
What is new is an overall awareness and knowledge on acoustics, we no longer accept a bad room as something inevitable, we actually do something about it to fix the problem.

Acoustic absorbers on the rear wall next to the S1.2 rear surround speaker

 

3. Measurement capability

Today any amateur hobbyist can set up a rig for acoustic measurements on the speakers and room using very little money, and the performance will be equal to or in many ways much better than what you would have paid an amount similar to a new car to get in the good old days.

You need a portable pc - which you already got, microphone, I/O interface for the pc, and the free software measurement package REW.

This will allow you to actually see what is going on when you try to set up your system, save you a lot of time - if better sound is the goal, and probably kill some myths.

When you now are able to measure, you can go on with acoustic improvements and integrate you new high-efficiency speakers that rolls off everything below 60Hz to your new bass system.
Just being able to integrate the bass with the main speakers so that it doesn't sound like the typical speakers-with-subwoofer will be a very significant improvement.


PC laptop computer with REW measurement software running

 

Microphone on stand measuring the S1.2 surround speaker

4. Bass systems

Huge bass systems pop up everywhere.
Large horns, or multiple subwoofer units with large-displacement drivers, or dipole variations.

This is a difference that is very audible, very visible - they are large, and physically perceptible.
The home-theaters showed the way with large-capacity subwoofer systems with low frequency extension well below 20Hz.

Today, any serious audio system has a capable and properly integrated bass system. 

5. Scams busted

In the 90's a trend towards seeing audio as something more spiritual and supernatural really took off.
Established science and methodology were replaced by occultism and religion.
This was boosted by scam manufacturers appearing on the market with expensive cables and all sorts of products claimed to improve sound but having no technical foundation at all.

They are all busted.

Yes, the manufacturers still live on, they continue to spread their religion - often supported by a cult of believers, but today anyone can search around the internet and find information on those subjects and find the truth.
We know why the scam marketers continue to spread the word, we know why people hear differences that does not really exist, and there is scientific documentation on technical issues and listening tests.

If you encounter a salesman telling you a more expensive piece of wire will sound better, then it's either a scam, or the salesman is incompetent.
I would not deal with either.

6. Audibly transparent

It is apparent for anyone with a technical background that a piece of wire can not change the sound the way some people claim to hear.
This led to a better understanding of the concept expectation bias - what you see affects what you think you hear.
The next logical step then is to question whether other parts of the audio chain, like amplifiers, could also be so good, that they do not change the sound in any way that can be audibly perceptible.

This gives birth to the term "audibly transparent" - a device that is so good that it does not change the signal in any way that is audible.

Blind listening tests, often performed as what is called an ABX-test, show that indeed, the audible differences between CD-players, amplifiers, DA-converters are not anything like what you can read in the hifi-magazine reviews.
In fact, those controlled listening tests show that well constructed high-quality equipment does not change the audio signal in any way that is audible.

Today it is easy to establish whether a device is audibly transparent.
You measure it.
When all changes made to the original signal is well below threshold of hearing, it is audibly transparent.

Basically, everything up to the speaker terminals can be made audibly transparent.

Take a look back at the amplifier sound quality article - the test samples are still available.
Can you hear a difference between the power amplifiers tested?
Can you hear a difference between the original music sample and the one passed through the AV-receiver 4 - FOUR - times?

A consequence of the fact that one can actually use a AV-receiver in the signal chain and be sure that it is completely audibly transparent is better sound for less money.
Less money because they are mass produced and targeted at a slightly different market, and better because you now have signal processing capabilities like bass management and delay setting for all speakers so the subwoofer can be properly integrated.

But be very aware that this does not mean there are no differences in functionality or performance - output power of amplifiers must be properly dimensioned, and differences in functionality can make one device more suitable than the other.

..

In addition to the mentioned points, there is the difference in how we receive and play source media content.
Streaming from internet, pc-playback and the removal of physical media like the CD.
In this world the old vinyl records have their renaissance, and is now known as - vinyl.




Computer playback of music files. Kodi media player. 

If you are in Norway, you can visit the Kvålsvoll Design AS demo room to experience some of those new trends in sound.
You are welcome.

Testing Amplifier Sound Quality

This is something I have though about a long time now - finding a method to reliably verify sound quality differences in power amplifiers.

Finally I have set up the necessary instrumentation to try out the method in practical experiments.

Purpose

Today good sound is easy to achieve, most of the audio playback chain can be made sonically transparent with relatively little effort and money spent.

However, through building and designing amplifiers myself, and measuring and listening to many different amplifiers, I have got this controversial idea that power amplifiers can actually sound different.

Obviously they sound different if you push beyond power limits, and also amplifiers can be made so good that they are transparent for any source material.

But where is the limit, how good does it have to be, and what measurements can reveal the transparent ones from the only-good ones.

Method

The problem with listening tests for verification is that it is time consuming, unreliable and difficult to do proper blind testing of amplifiers.

If you could test using a software abx-tester, like the one in foobar, the task would become easier.

By recording the output signal from the amplifier while playing at decent volume, you get a sample file that can be compared to the original in an abx software player, and you can do the comparison on any playback system - headphones, speakers.

The playback will of course be affected by the amplifier in use for playback, but considering descriptions of the seemingly huge sonic differences described in reviews, it should not be a problem to hear at least some of the sound characteristics from the test object. 


Method for amplifier abx-testing; The blue boxes are sound data files, the green boxes are test instrumentation and should sound transparent, the red box is the amplifier being tested: 

Verifying the recorder

After rigging the recording system, the first task is to verify that the recorder is sonically transparent, or at least much better than the amplifiers I want to test.

The first recorder system is made easy and practical, thought I could test this first, if it fails I have some options to improve it if necessary.

I recorded some samples from the output going to the amplifier, so that I can use abx to verify if there is an audible difference from the original sample to the recorded sample - with no amplifier in the loop.

I also took one of the samples and sent it through this loop 5 times, as I suspected the differences would be too small to be detected.

If the 5x loop sample still sounds good, then the recorder can be assumed to be good enough for the purpose.

After several attempts trying to get an abx with positive outcome, I conclude I am not able to hear any difference from the original and the 5x looped sample.

I used two different set-ups with different speakers and amplifiers, and I used headphones.

It might be interesting to observe that the recording loop includes the DAC and pre in a commercial avr.

To test that the method actually can detect differences in sound quality I encoded 128K mp3-samples and compared those to the original.

All mp3-files tested positive.

The instrumentation is verified by abx testing of Instrumentation for playback and Recorder - the green boxes:

 

Amplifier testing

I recorded 3 amplifiers, they are quite different in topology and also measures different, though amp 1 and the C15 both measure very good on parameters assumed to have significance for sound quality.

The problem now is that I have a very hard time trying to hear any difference between any of the amplifiers and the original sample.

I uploaded sample files to the web site, they should be accessible by typing this link:
www.kvalsvoll.com/Articles/abx/

They are tagged with a very brief descriptive title.

Start by looking at the tuttabella_test files, do an abx of the mp3 - it is harder than you think..

The sample files are music and signal samples, in original version and sampled from the loudspeaker output of the amplifiers.

Amp 1 is a very good commercial amplifier.
Amp 2 is a low-budget avr.
C15 is a 15W design by me.

The music samples are chosen to cover different genres and style, but also for ability to reveal differences.
Lots of high frequencies and voices in the Tutta Bella should be good for this purpose, the Stinky and Humming samples are more electronic and jazz-funk style with dynamics and full frequency range.

The 19K+20K signals reveal high-frequency intermodulation.
If you can not hear above 19K, all you hear is distortion, younger people may find it very annoying.
The distortion will sound like high-frequency hiss, and a tone at lower frequency, more specific 1KHz.

The 4K+10K+16K is meant to reveal high frequency distortion, mainly intermodulation, while the distortion-generating tones are also clearly audible.
The distortion will sound like high frequency hiss, and a metallic tone.
The tones are very annoying to listen to, and you will need to listen quite loud to have any chance of hearing any distortion.

WARNING:
Some of the files contain signals that may or may not be audible - DO NOT TURN UP THE VOLUME BEYOND NORMAL LOUD LISTENING LEVEL.

You may end up destroying tweeters or amplifiers, be careful.


Update 20.03.2015:
All music samples were recorded with the amplifier playing at a decent volume into a real loudspeaker load - just like you would normally listen to music.

Level was set to -3dB below clipping for the small C15 amplifier for 0dBFS from digital source material. This equals a peak level of 10W into a 8 ohm load.

Not conclusive

I have not yet managed to get a positive result for the music samples with amplifiers.

I made the im_* test signals to see if some very nasty test signals works better.

And they do - I can verify a difference between amplifiers and original for those signals, which is interesting, because that indicates a possibility for audible differences in amplifiers.

Further experiments could be to record amplifiers looped several times, a 4x loop will increase nonlinear distortions by 12dB, while noise only increases 6dB.
If this is enough to clearly reveal differences in sound, it may be used to learn and train hearing perception because you now know what to listen for.



Update 23.03.2015:

Sample files updated due to an error on all 44.1K sample files, the error is audible on the multitone samples im2, im3.


Music samples are now gain matched within 0.1dB.

Additional samples from instrumentation test loop added, these are labeled xxx_test. 
The 5x loop test is removed, now replaced by an updated 4x loop test.

It is recommended to update your sample files if you already have downloaded any previous version.

Center distance adjustment done correct

Learn how to set the center distance distance correctly to get the best sound.

Center distance setting

The center distance is a setting in the processor/AVR, affecting time delay of the center channel relative to the other speakers.
The value is set by automatic calibration, or manually, to the physical distance from the listening position to the center channel speaker.
We will see how more accurate fine adjustment of this distance can improve sound quality and how to do it.

Effect of center distance

The distance setting affects the timing of the signal to the center, and thus the summation of signals across the front stage.
If the three L, C, R front speakers have different timing, the summed frequency response will not be correct.
In particular sounds that are panned across the front will be affected.

If the phase response from the center is different from the L R mains speakers, it will not be possible to get a smooth total response.
The concept of "timber matched" speakers is quite misunderstood, as it is not the typical small differences in frequency response amplitude that causes problems, but differences in phase response and off-axis radiation.
The center speaker must have a phase response and radiation pattern similar to the L R speakers to be able to provide seamless front stage sound.

If the distance is far off, more than 20cm, the low frequency response will be affected.
Smaller errors down to around 2cm is perceived as loss of focus and clarity at higher frequencies, for signals spread out on all front speakers.
For movie dialogue recorded only in the center channel there is no difference, even for large deviations.
So, for those hoping this is a way to move the center sound image further away from the listening position, to make it more similar to using a phantom center, unfortunately that does not work.

How to verify center distance

We can use a correlated pink noise signal played through all front channels and monitor the frequency response to see if the channels sum correctly.
Wrong timing will show up as comb filtering, visible as dips in the response.

How to adjust center distance

Start up your functional REW or similar acoustic measurement set-up, and open the RTA real-time analyzer.
We will use correlated pink-noise for measurement signal:

Correlated pink noise signal for LR
Correlated pink noise signal for LCR

You can create your own correlated noise signals by first making one channel pink noise, then copy this signal to the other channels.

Frequency response graphs shown are from RTA with settings for quick and reasonably correct presentation of the situation above around 1KHz, at low frequencies the graphs are not accurate.

First, we must align the microphone exactly between the front L and R at the listening position.
Do this by playing a correlated pink noise signal in L and R, and move the microphone sideways until the response is most flat at high frequencies.

Mic alignment: Initially, the mic is far off:

Mic alignment: Getting closer:

Mic alignment: After fine adjustment within approximately 5mm:

Set the center channel distance to a reasonably correct value, within say 20 to 50cm.
Also check that the center channel level matches the front L and R, and adjust if necessary.
Then play a correlated pink noise signal in all L, C, R channels, and adjust the center channel distance in small 1cm increments until the response is as flat as possible.
It may show a slight roll-off at higher frequencies, as even small differences in phase response between center and L, R will have significance.

Center distance: Initally, we see the signals do not sum up correctly, there is visible comb filtering:

Center distance: After adjusting the center distance within +-1cm:

Subwoofer system integration

At low frequencies the integration to the subwoofer system is affected, and this is solved by adjusting the distance for the subwoofer system.
Subwoofer distance adjustment will be covered in a coming article.