Saturday, March 29, 2014

Audibility of peak limiting

Peak limiting and clipping is used to make music seem louder, but does it really work that way?
No - removing the peaks actually reduces impact and brutality and makes it sound flat and boring.


Typical sources of peak limiting in audio reproduction are loudspeakers and power amplifiers.
This can obviously easily be avoided by turning it down a bit, or get larger speakers with better sensitivity.

Music can also be destroyed in the production process, where removal of peaks using brickwall limiters is more common than not today.
Dynamic compression, brickwall limiting and peak clipping in music is well known today as the Loudness War.
The kind of signal processing used causes loss of transient peak amplitude and adds distortion.
It is this type of signal destruction that is investigated here.

Upon visual inspection one can see that the limited waveforms are clearly changed, and it may be difficult to understand why this is not also very easy to hear.
But the loss of signal amplitude happens in a very short period of time, it is not necessarily easy to detect unless you know what to listen for.
There is no apparent change in overall sound level, and the tonal balance is not changed at all.


Reduced transient peak amplitude will cause loss of tactile impact and punch when played reasonably loud through loudspeakers.


A music signal with very dynamic and transient content - mostly drums - is processed with a limiter to create peak limited test signals.
Three different test signals were made - original, -3dB limited and -6dB limited.

The original and the distorted signals are ABX-compared listening on headphones and on the big system.

Test signals



The less -3dB limited signal was difficult to distinguish from the original, and the difference was perceived as more noticeable when playing reasonably loud on loudspeakers.
The difference was noticeable also on headphones, the limited signal seems slightly flatter sounding on the loudest drum hits.
On loudspeakers there was a barely noticeable loss of physical transient impact.

For the heavily -6dB limited signal differences could be verified as audible even when not playing very loud, there was distortion causing a reduction of clarity and perceived sound quality.
On loudspeakers there was a clearly noticeable loss of physical impact.

ABX results, headphones:
Limited -3dB: Total: 8/10 (5.5%)
Limited -6dB: Total: 10/10 (0.1%)

ABX results, big system:
Limited -3dB: ABX: 16/14/0.21%
Limited -6dB: ABX: 10/10/0.1%


The results does not dismiss the hypothesis that peak transient amplitude is important for tactile impact and punch.

The less -3dB limited signal was difficult to distinguish from the original.
When looking at the waveforms the signals are clearly different, but when listening the differences are not so easy to detect reliably.
Even the -6dB limited signal could pass unnoticed if there was no original reference to directly compare it against.
Also, the reproduction equipment - mainly speakers - must be able to reproduce the transients without additional distortion.

The combined results from this limiting test and the phase distortion test indicates that it is possible for such errors to be introduced unnoticed in the music production process - they simply can not hear it.

The important part is that the severity of the transient distortion depends on how you listen.
When sitting down to really enjoy your favourite music, you turn it up, and that is when the lack of life and dynamics is most apparent.

Loud music production style utilizing heavy compression and limiting has several other negative effects on sound quality.
The negative consequences of lost transient impact investigated here is only one part of the destruction.

Tuesday, March 25, 2014

Audibility of phase distortion

Phase distortion at low frequencies causes audible degradation of music.
When is it audible, what causes it and how does it affect sound quality.


Phase distortion means that different tones gets a different time delay.
It has been a general assumption that phase distortion in audio reproduction is not audible.

Typical sources of phase distortion in audio reproduction are loudspeakers and room acoustics.

Music can also be destroyed in the production process, where dynamic compression can cause effects similar to phase distortion.
Dynamic compression in music is well known today as the Loudness War.
The kind of signal processing used causes loss of transient peak amplitude, and smearing of the same transients over time, to compensate for the amplitude loss.
It is this type of signal destruction that is investigated here.

Upon visual inspection one can see that the phase distorted waveform is clearly changed, and it may be difficult to understand why this is not also very easy to hear.
But phase distortion means that the only change to the signal is related to timing, the spectral frequency distribution is the same, and there is no nonlinear distortion added.
The energy of the signal is preserved, though very large phase shifts will cause the energy to be smeared out across a longer time interval.

One thing that may be important is that the transient peak amplitude is reduced in level, and this is what is presented as a hypothesis for audible difference here.
This reduction in peak level will potentially reduce the maximum peak sound pressure that is experienced.


Reduced transient peak amplitude will cause loss of tactile impact and punch when played reasonably loud through loudspeakers.


A music signal with very dynamic and transient content - mostly drums - is processed with an allpass-filter to create a phase shift that is large enough to change the signal so that the peak level is significantly reduced and the waveform is visually changed.
Three different test signals were made - original, phase distorted, severely phase distorted by running all-pass twice.

The original and the phase-distorted signal is ABX-compared by listening on headphones and on the big system.

The big system is a full-range system with reasonably flat phase and group delay through most of the bass range.

Test signals: Original, 1x allpass, 2x allpass


Listening on headphones, the phase distorted music could not be detected as different.
Only when comparing to the severely phase distorted signal could a difference be verified, though now the differences were very clear as the distorted signal sounded more boomy and smeared in the bass.

The phase distorted music signal could be verified as different from the original in the ABX-test, on the big system.
The observed subjective differences are that the phase distorted signal lost some impact and punch.

ABX results, headphones, allpass 2x:
Total: 10/10 (0.1%)

ABX results, big system, allpass:
ABX: 16/14/0.21%


Phase distortion at low frequencies is audible and can cause a significant degradation of music, in certain specific situations, such as when listening on high quality loudspeaker systems at fairly loud levels.

The results does not dismiss the hypothesis that peak transient amplitude is important for tactile impact and punch.

In other situations, such as when listening on headphones, or at lower volumes or on lesser capable speakers, it was not possible to detect any audible difference between the phase distorted and original signal, as long as the added phase change and time delay is within reasonable limits.
This also explains how it is possible that such errors are introduced in the music production process - they simply can not hear it.

Next up

A similar test for peak limiting is coming up soon - is it audible, and how does it affect the sound.

Tuesday, March 4, 2014

2-channel with subwoofers set-up

I present the set-up of a 2-channel system, with simple subwoofer calibration.

With measurements shown, so that we can see what is actually going on.

And I will show how to try without measuring, only using a pink noise signal that can be played back from a file.

This is a standard 2-channel system with digital or analog source, controlled by an analog preamplifier.
No DSP, no room correction, no bass management.  
The subwoofers have no digital processing, the only adjustments are level, crossover frequency (low-pass filter) and phase.

The main speakers will play full range down to whatever frequencies they can output, there is no low bass cut-off. 

This configuration is what many enthusiasts actually will have for a 2-channel system.
Now I will show that it is possible to integrate subwoofers into such a system with great success, even though there are limitations and compromises will have to be made.

The addition of capable subwoofers transforms this system into full-frequency range with reference quality bass.

This is a nice, smaller system, with satellite main speakers driven by a class-A amplifier with low output power.
Max SPL is rather low, we hope for a nice and pleasant sound with a true three-dimensional presentation and great bass with precision, slam and capacity all the way down to far below hearing range.

Summary of calibration adjustments:
  1. Placement not too far from the main speakers, but try to get positions with good gain and smooth response.
  2. Adjust phase individually on subwoofers to get the best response of subs only.
  3. Low crossover, 60Hz, max 80Hz.
  4. Use RTA (Real-Time Analyzer) to see effects of adjustments in real-time. 
  5. Adjust crossover and level on subwoofers to get as close to your target response as possible.
  6. While having the measurement gear out this is a good time to try alternative positions for the main speakers.


The important parts for this exercise are the main speakers, subwoofers and the room.
Source and amplifier is not that important for set-up and calibration, as the process and measurements will be the same for any amplifiers without processing for crossover and delay.

3.6m width x 4.5m length.
Door opening on right wall, large opening into other room through rear wall, window on left wall.
Other room length 4.1m together with this room defines lowest length node at approximately 8.6m.
Wood panel on all walls and ceiling, wood floor.
Some chairs and a round table, media console for audio equipment.
No additional acoustic treatment of significance.

Low frequency response of this room is dominated by the total length of both rooms and the door opening acting as the port in a resonator where the room is the cabinet. 
This causes resonances from around 15Hz up to around 25Hz, where room response is strong. 

Main speakers:
Small satellite system with 8" bass driver sealed, 2x 14cm mid, 1" dome HF.

2x small Compact Horn subwoofers with 10" driver (105dB/20Hz/2pi/1m, equals approx 12" ported or 15" sealed of high quality).

Measurement instrumentation:
REW measurement software, calibrated microphone, computer, USB I/O.

Initial placement

We start out with the system set up like this:

Room with L+R speakers and subwoofers initial placement

We want 2 subwoofers to be able to get a reasonably smooth response.
That also gives more output (+6dB), so that we can use smaller subwoofers, it is easier to place 2 smaller boxes compared to one giant monster.

The subwoofer locations can be determined by measurement, either by measuring at the listening position (LP) while moving subwoofers, or by placing one subwoofer at the LP and then move around with the mic.
Use the RTA in REW for this, run Pink PN Noise, then you can see the response in real-time while moving the mic around.

The fact that we have no way to adjust for time differences between mains and subwoofers will limit placement options.
Ideally the subs should be a little closer to the listening position (LP).
Here we start out with a practical location that also works; in front of the main speakers, towards the corner.
Bass is strong, and because the sound radiation is from the front of the subwoofer, it is radiating at some distance away from the corner, which usually gives a smoother response.
In a set-up where the best sound quality is the goal, we do not want the main speakers in the corners, so this location is free to use.

Placement of one subwoofer towards the rear of the room, behind the listening position, can also be an option.
What will work in your room will depend on the acoustic properties of the room, as well as obvious practical considerations.

Initial measurements

We measure the main L and R speakers from the listening position (LP):

Frequency response L+R no subwoofers initial position

And the 2 subwoofers:

Frequency response subwoofers before any adjustments

Sub phase adjustment

First the phase is adjusted on one of the subs to fill the response dip around 50Hz.
For this particular room and sub locations this adjustment makes a large difference, but that may not be the case in a different room.
Obviously, the room is not acoustically symmetric for both sub locations. 
Use RTA, adjust while monitoring the response in real-time.

RTA running for subwoofer phase adjustment

Frequency response subwoofers phase adjusted

Then the phase on both subs are adjusted so that they fill in with the main speakers.
Use RTA, watch how the response changes around the crossover frequency, adjust to max level.

Frequency response L+R+subwoofers initial position

Sub level and crossover

Adjust to achieve a reasonable frequency response that extends the natural downwards tilt, and you may prefer some extra level at the lowest frequencies below 40-30Hz. 
Consider choosing a crossover rather low, below 80Hz or lower, to minimise effects from time alignment issues.
Note that it may be necessary to adjust phase after changing the crossover.  

All this is best to do with the RTA, so that you can see the changes in real-time.

Adjustment without measuring

Using a pink noise signal and your ears to adjust phase, level and crossover for subs may at least give some improvement compared to no adjustment at all.

Play a 200Hz bandwidth-limited pink noise signal from file.
Adjust phase on one sub for loudest possible signal.
Then adjust phase on both subs to get loudest level around the crossover frequency.
This is the more difficult part, because is is not easy to know how the noise should sound when the setting is correct.
Level and crossover is not easy to adjust using pink noise, because you would have a problem knowing how it should sound when everything is right, unless having a reference readily available to compare with.
Headphones may be a solution - listen on headphones, then compare to how it sound on the system and adjust to same sound.
It may be better to use some known music with bass, as actual music is more likely to reveal when something is way off, compared to a noise signal that still sound like what it is - noise.
Obviously this is something that is not very easy to get right without measurement.
The result depends heavily on the chosen program material and the listeners ability.

Pink noise signal files can be downloaded from Kvålsvoll Design here:

Pink Noise 100Hz limited
Pink Noise full bandwidth

L+R position

Seems like the tilt is a bit too much down at higher frequencies, and it also sounds rather dark with recessed highs.
In the midrange the response could be more flat, this is likely caused by boundary reflections from walls and floor around the speaker.
So, we try to relocate the speakers a bit closer to the listening position, making the distance to the front wall larger.
The larger distance between the speakers and the front wall also usually improves imaging and perceived clarity in the midrange.
The speakers are also raised around 10cm to bring the tweeter on-axis.
Angling is adjusted so that on-axis points exactly towards the listening position.

L+R speakers moved to new position closer to listening postion and raised

The response now looks much better in the midrange, and the tilt is reduced.

Frequency response L+R+subwoofers position adjusted

Frequency response L+R+subwoofers before and after position adjusted

The impulse response is surprisingly good, especially considering the type of speaker and no acoustic treatment in the room.

Impulse response

The step-response can be used to see if the subs are properly time aligned with the mains.
Well, they are not, but that is not possible to correct when we have no delay adjustment on the mains.

Step response 

The group delay also provides useful information about timing issues.
If subs and mains are properly aligned they will end up in-line with the same average group delay in the area around the crossover.
This can be difficult to see because this frequency range is usually affected by room and boundary reflections causing large group delay deviations.

Group delay

Checking subwoofer capacity

We check the output and distortion of the subwoofers to see if it is likely that they can match the main speakers.

Distortion measurement sweep show 110dB at 20Hz with 1.7% THD

Indeed, the small Compact Horn subwoofers easily outperform the mains with 110dB at 20Hz with 1.7% distortion.
They can obviously do a lot more, I did not bother to check, as that requires recalibration of the mic amplifier for more headroom.

How does it sound

The subwoofers seem well integrated, providing full-size bass with a heavy bottom, both impact and weight, dry and powerful.
It does not get boomy or muddy in the bass even when the bass is running a little hot.

The relocation of the mains further away from the front wall improved perception of depth and clarity in the lower midrange. 

The biggest problem is limited output capacity, it simply can not play loud.
The bass range is not as smooth as one can experience from the best systems, and we know from the measurements that the response is not entirely flat.

Further improvements

The bass is quite heavy at the lowest frequencies, level is low around 50-70Hz, and there is a peak at 100Hz.
This can possibly be improved by moving one or both subwoofers, and adjust subwoofer level and crossover.

Since this set-up was done primarily for this article, I am satisified for now.
If anyone in the neighbourhood wants to hear and experience this system please feel free to contact me.

If you are interested in reading more on audio systems and set-up, check out my article How to  Set-up a Home Theater System.

Sunday, March 2, 2014

Floor reflection fixed

Floor reflection fixed with two small interior add-ons

Puff ready for some music

When measuring speakers I have used cardboxes, carpets and such in front of speakers to dampen the first floor reflection.
Useful both for indoor and outdoor measurements, this proves to at least have some effect on the reflection, improving measurement accuracy in a quick set-up.

Now, could a similar approach be used in an ordinary room, and actually look nice and appear as a natural extension of the interior design.

In The Moderate Cinema the main front speakers live in a very tight space, with lots of things around them, causing reflections that affect the response especially in the midrange.
Normally, one does not treat the floor reflection, it also causes practical issues in most cases.
However, in a room like this it may prove to be more practical to fix the floor reflection than trying to do effective damping and necessary relocation of other items.

Initial experiments with card boxes covered with pillows and blankets show that this actually is a good idea, the frequency response is significantly improved.
This can be made as a moderately sized puff, or footstool.

Frequency response initial testing - grey: bare floor, cyan: cardbox and pillows

The completed design is a puff measuring 38x38x30cm, made of ordinary furniture foam covered with furniture fabric, and an optional matching blanket or pillow on top.
Looks nice, is not too large, integrates well as part of the interior design.

Foam body for just the right acoustic absorption

Completed puff

Carefully placed in front of each main front speaker, they significantly improve the frequency response and the impulse response in the midrange.

A location somewhere around the middle between the speaker and the listening positon will often work good. In this room there is a table in the way and the best locations are somewhere rather close to the speakers. 

Puff installed

Frequency response shows reduction of peak 700Hz and dip at 400Hz in midrange

Filtered IR at 1000Hz also shows improvement 

The group delay shows reduced reflections in the midrange

The full-range IR show no significant difference, not so strange due to the very controlled directivity of the loudspeakers at higher frequencies