Making the listening test

After giving it some thought, I am going to normalize all the clean signals to -10LUFS and run the listening test at 80dbA as suggested in ITU-R BS 1770 -3

I will then have listeners change the level of a distorted file, to be the same.

This is a similar format to what the ITU did when testing LUFS for multichannel comparability.

It will also mean data can be compared fairly.

If we and LUFS are correct, the listeners should choose a level where the LUFS of the distorted signal is negligible to that of the audio signal.

A less distorted signal will be boosted more and a more distorted signal will be boosted less.

There is good evidence that LUFS is good stuff, when looking at :

The Effect of Dynamic Range Compression on Loudness and Quality Perception in Relation to Crest Factor

This not only suggests that LUFS is pretty accurate, but that a change in crest factor using dynamic range compression will effect loudness perception

Therefore, we can compare level coefficients to correlation, THD, crest factor, harmonic content, dynamic range and LUFS.


Even if the listener boosts the level of the distorted signal beyond the level of the clean signal, this will show that there is an effect on perceived loudness. 

I am going to wittle down the levels of the clipper to reduce the THD caused to a 1kHz sine wave to roughly 0 - 20% THD. This is because beyond this level, the distortion is very audible and I want to minimize the subjective effects of distortion on results. 20% THD is audible enough as it is. A further study could look at those levels of THD. Other studies previously referenced have proven that distortion becomes audible beyond 15% THD, when compared with the effects of clipping on a pure tone.

This will also allow me to slip the signals through other nonlinearities which function with frequency, and see how the different harmonic patterns compare.

I need to learn how to make waterfall plots for these signals. Maybe spectrographs will have to do?

Anyway.

Nonlinear dynamics

https://www.youtube.com/watch?v=Y0u3o9_BRVQ

Anyway, it has been a little while (i think) since I updated this blog.

Stuff is moving along well. The listening test is due for a slight modification.

Next up I am going to apply the LUFS measurement algorithm in matlab, and use it to process the loudness of all audio files used, to ensure a solid LUFS level.

I will then standardise the listening conditions to a particular LUFS measure using a dumy head and the listening test rig, to ensure everything is the same. I can then increase the LUFS on the test rig to test different loudness effects on the effects of DISTORTION on perceived loudness.

I cannot directly link only THD to perceived loudness in musical signals, so I am just going mainly for a distortion characteristic now,

Also, I will evaluate crest factor and other bits with the data I accrue to get a more accurate picture of the systems involved.

Finally I will perform some cross-correlation analysis on audio that has been passed through nonlinear devices, to give a number to amounts of distortion, even if it isn't a traditional distortion metric. I am also modifying the listening test to get relative loudness measures between a clean and distortion version of a signal, so we can find what the level change is for perceptual level difference.

Good stuff.

It is just a shame that I have no listening test data yet.
I am also thinking about simplifying the listening test, to make it less prone to error.

Update

A newer version of the listening test has been completed.
This takes the 7 samples at different levels of clipped distortion (known THD% conditions for 1kHz sine), and allows users to listen and choose preference.

The paper of the day is 'The Correlation Between Distortion Audibility
and Listener Preference in Headphones' by 'Temme et al', which basically showed that there is no direct link between distortion audibility and listener preference in headphones. It also warranted that more studies on a perceptual basis need to be undertaken, and thus my paper will be obsolete by the time those occur I expect.

It does give the go-ahead to set up the generic test pattern for data sets that I had planned, because there is not a defined standard set of parameters out there as yet. Although the ITU-R 1770-2 method for setting headphone loudness should be considered to make a fair test.

'The hawksford paper' doesn't reference a measurement standard for the listening tests, which make a good comparison hard to do, in relation to their method.

Two more things. Both backup ideals for test measurements.
1: randomly generate some waves with frequency x, and get listeners to pick which nld sounds loudest, and which one sounds clearest.

Minor Brainwave

The problem with my originally proposed method may have been convolving with a sine wave that does not have a DC offset? Because of 0s...

Idea

Maybe this is total nonsense...
What if I found the THD% of a nonlinear device for a bin of frequencies...
I then convolved that behaviour with the Frequency domain version of some music.
I would then have a signal with a known average thd% per f bin.

I can then add and remove that harmonic content as a function of a slider, and know the THD%av. The thing to remember is that, if I add x much energy over the frequency response at that frequency, and then reduce that by half, that is still a known amount of energy added. Der Untergang is the IMD and other artifacts possibly caused by adding this content. That said, an amount of energy + an amount of energy is still that given amount, in this instance. I would no longer be applying the NLD to the signal, just its effects and thus I may avoid some of the other effects of the NLD on a boradband basis.

EDIT: Reality check

Some of the fundamental properties of signals.

So a signal may have multiple channels, but each channel will effectively be just a set of number values through time (in matlab). This change of numbers will infer spectral content throughout a time period, but will in effect just be an intensity value at any single point. in time.

Continuous Time & Discrete Time

In continuous time, this would mean that THD should only be there for the response time of the device in the instance of an impulse, and by extension the THD pattern will change with continuous time. This does not have to imply IMD, as the distortion characteristic (DC) of the device should imply the DC output with signal instantaneously (assuming it is a memoryless device such as those that I am using). The problem may become more apparent, when taking time variant devices into account such as capacitors and inductors.

So if you chop the signal into small packets of time, the same must be true?
This is why the signal must be processed in the time domain. If you did it in the frequency domain it wouldn't work.

If you were really clever, you would find a way to separate out the THD effects over time, and see the difference, sample to sample. There must be a way to take an NLD, and simply get a numerical output for the distortion value, without the music? Because you loose frequency information when you look at small segments of the time domain.

Here is a concept for you

Is it not possible to comprehend that the difference in sample time, may allow for time variable devices to correct themselves, if you placed a small enough get in-between audio samples. The gap would have to be pretty small, or in a digital amplifier.

After a couple days to think on it, I think that a THD% study using music will not work.
Why?
Because signals with multiple frequencies at once, in

How very convoluted

So the next step in the process I have deemed, is to find a way to quantify the THD of an audio signal. As it turns out, this is a hard thing to do. I have been trying to do this quite a lot.

In fact, a lot of the papers I have referenced so far (mostly in the proposal) are based on this particular point. The reality is, the original question is in itself questionable.

THD% is a way of measuring distortion. By definition, distortion and complex signals cause more distortion i.e. inter-modulation, and various other properties of non linearity.

The question shouldn't really be 'does total harmonic distortion have an effect on perceived loudness', but more likely 'does amplifier/loudspeaker distortion have an effect of perceived loudness'. These are not the easiest things to determine in some cases, but it would be a much tidier question all in all.

I have most recently been trying to use a home-brewed method to calculate the THD of a music signal. Convolving a 1kHz sine wave with the distortion characteristic of the music which has been processed with a nonlinear device. The problem I am having is that a 1kHz sine is spectraly sparse, and most of the distortion seems to occur in a wide band at high frequency.

Key papers at this time are:
Measurement and Perception on Nonlinear Distortion - Comparing Numbers and Sound Quality
- Alex Voishvillo
Testing Challenges in Personal Computer Audio Systems
- Wayne Jones et al.
Nonlinear Distortion Measurement in Audio Amplifiers: The Perceptual Nonlinear Distortion Response
-Phillip Minnick
Auralization of Signal Distortion in Audio Systems Part 1: Generic Modeling
-Wolfgang Klippel
Non-linear Convolution: A New Approach for the Auralization of Distorting Systems
- Angelo Farina

One of the biggest problems I am currently having, is that if I re-analyse a pre-distorted and analysed sinewave, I get a different value to what I can clearly see on the graphing of the function.


So it is time to step back, look at the question and make a decision.
Do I forge ahead and stick with THD as the metric for distortion measurement, or do I go freeform? I think the NLDs cause IMD and all other kinds of goodies for analysis, and convolution is going to become a big part of getting real answers from the complex program material. In the mean time I am going to see if I get start to get my understanding correct in relation to 2 sinewaves, some impulses and some noise.

I must also look towards perceptual models, in case analytical ones do not work out.

Listening Test Beta version

The aim for this weekend, was to get a listening test software beta together.

A deliverable for John, to show that I can deliver. 

Take into account that before this project, I had never touched Matlab.

Well, once I did, using a program Adam had made, I entered data and it spit out a polar plot.

But really nothing of any use relating to this project.

Last night I completed this first working beta version on my listening test software. 

There are still some things to change i.e. how it exports test data and biasing the sliders to the middle. Otherwise, it is just a bit slow and clunky. This is because every time you move a slider, it recalculates the mix of tracks, adds them and normalises them before playing from the same sample. Initial listens show some very strange, interesting and unpredicted results. As such, I need the software to be reviewed by someone who actually understands how Matlab works to a good level. 

My advances have been great in a very short space of time, but there is still a mountain to go!

So here the is format of the listening test as far:

The listener has to go through 7 genres of music, listening to each track at least twice. The listener for each track, is to move the appropriate slider, one to find the best sound, and once to find the loudest sound for each genre. The listener is also to put in their sex and age, and save the answers between each genre. I haven't finished how saving and exporting data works yet. For this, I need Adam or Bruce.

It has been a learning curve, the last 24 hours. But here it is.

I aim to optimise it better, change the save behaviour, and make it much prettier. But it is a deliverable beta, put together in roughly 10 hours.

I am pretty chuffed.

The data recorded is fully quantitative, in that numerical values between 0 and 1 are chose, by listeners who don't know what they are listening to i.e. without deception by still blind.

The next step from a data perspective, is to polish up the sample choice, and look at the crest factor of each sample at given times in the sample. Then once listening test data is back, determine the difference in crest factor for the highest and lowest populated data points. 

I think it is in fact the effect on crest-factor as I originally thought, that determines perceived loudness in these basic listening tests. I believe that greater THD makes lower mid and bottom end more perceptible, by masking the top end clarity of the music. I believe that this perceived extension in bass is what clubbers want, and you can hear it in the sample variations when listening to a 2.1 speaker system. I haven't tried headphones yet. I think the next step is more tests with the better quality samples, and multiple NLDS.

Really, testing needs to happen with people whose ears have gone into 'protect mode', with the 3piece system seizing up a little. Unfortunately, there is no way the ethics police will allow that. So I may need to look into modelling that mathematically. I will also need access to whatever the military has on this behaviour.

My aim by the end of the week is to have beta V2 together, and to have done a reasonably hardcore literature review on crest factor and the ears internal protection system. 

Woop! 

Below is a link to the folder including the Matlab embed of my beta.

In the folder above, the first test interface is here.

Houston, we have a mild breakthrough

I have started to develop the listening test interface.
It involves using a home built user interface, and functions written in Matlab.

The function works using 'audio player' objects, and does an iteration of the code every time a slider is moved. The problem is editing large audio files makes the program slow and clunky.

Listeners will compare two versions of a bunch of tracks, and decide how much distorted track they like in the mix.

I have borrowed an MBox 2 from the university, and have bought a new set of headphones for these tests.

Getting the % effect and what percentages I am aiming for

It came to me in a dream...

Remove the original audio from the induced distortion, normalize to a percentage of the level of the original audio signal and add the distortion to the original signal again.

Well, this was mild nonsense. It wouldn't have fixed the THD% calculation for music, because there is already spectral content outside of the fundamental. Therefor, the calculation cannot work for music in the iteration I am using for this project. This was something Adam and I spoke about in the first place.

The new plan (which was really the old plan before the tangent), is to modify quantities in the nonlinear equations, to give a distortion % to the 1kHz sine tone that everyone uses (normalized to 1). Then apply that filter to the audio samples and the different percentages and have some samples for testing.

These will be ready by tomorrow (with luck). After speaking with Adam, I have decided to vito the rectifying NLDs because they aren't similar to a normally working device in saturation. Although, interestingly, squaring the wave pitch shifts it if it is a sine wave. The behavior to other wave types is even more interesting. I would love to start looking at distortion characteristics in deeper detail for post-grad.

Back to the scheduled program.

To decide what percentages of THD are worth exploring, I fist looked at the Geddes & Lee paper 'Auditory Perception of Nonlinear Distortion', which discussed a new method for measuring the perception of distortion (THD, IMD, ETC) in listening tests. This was all good, but I wanted to see what else had been done before I commit to that method. I looked through 'Nonlinear Distortion Measurement in Audio Amplifiers: The Perceptual Nonlinear Distortion Response' by Phil Minnick, which pointed me towards the Voishvillo paper 'Measurements and perception of nonlinear distortion - comparing numbers and sound quality'; which . There was also 'a new method for measuring distortion using a multitone stimulus and non-coherence' by Temme and Brunet.
Got that?

Finally, I came to the paper 'Measurement of Harmonic Distortion Audibility Using a Simplified Psycho-acoustic Model - Updated' by Temme, Brunet and Qarabaqi. This paper suggested that distortions that THD% samples between 0 and 9 had a lower standard deviation of subjective grading, where as at 10%, the deviation of this grading became much wider. These ratings were from 1 - 5, good to terrible. At 1-% some ratings jumped up from below 2.5 to above and up to 3.4. This is in line with a number of other papers, which suggest that THD becomes more perceptible above 10%.


As such I will use samples at THD%s in steps of 0.5% between 0% to 20%, and then steps of 10% up to 80%. This is because above the threshold conscious of perception of distortion, the listener will know they are hearing a strongly distorted signal (as proposed in the results of the paper mentioned above), and therefor the accuracy of the effect may be less.

The order of test signals in listening tests will be randomized, to fit in with the methods I will discuss in the next post. "I got it rapped like a mummy", Dr Dre.

From the hawksford paper, I am going to focus on the Cubic/SQS NLDs, as they may follow the behavior of saturated amplifiers/loudspeakers more readily.

Non-linearity

So right now I am battling with the linearity of nonlinear equations.
In the Hawksford paper I linked in the last post, there are a number of nonlinear equations.

This includes half-wave rectifiers, full wave, SQRT, Clippers (the obvious one), Fuzz exponentials and other bits and pieces.

The difficulty I am currently facing is that, by extension, some of these nonlinear equations are behaving constantly, regardless of the amplitude offset I give the normalised signal I feed them.

This may seem dead obvious, but I think I am about to start venturing into unknown (to me) territory. For example: applying the absolute square root of a signal to itself seems at the moment to be causing a constant 138.5332% total harmonic distortion to the signal, regardless of the amplitude offset I have given it. I think the problem is not in the equations, but my approach to signal modification. At the stage of distortion, the signal has already be normalized and split into mono channels for processing. I am sure the behavior of some of these nonlinearities is very particular in terms of a bandwidth of amplitudes, and even though everything is normalized to one, there should be a bit more variation in THD%. A fundamental problem with what I am doing is comparing the THD caused to a sine wave, with the effect the NLD is having on some music. How can I change things and expect the results to be constant?

More paper reading to be done this afternoon for some literature review. Maybe I will realise the cripplingly obvious thing I am missing. Either way, I am on schedule at the moment.


My aim is the apply a definite and controllable % of THD to a signal, by increasing and decreasing the amplitude of the signal. I need to come up with a way of pre-determining for each nonlinear equation, what ratio of the signal sits above a given threshold (in amplitude), and modify said threshold to give how much a signals total level needs to be modified by to achieve the given % THD.

Looking forward to the listening test, I think I will have to make it loud enough to force everyone into the acoustic reflex (middle ear compression), while still sitting in CONAW levels. Its a point of compression that fundamentally happens at different points for everyone. I might have to start thinking about more ear based factors too, like masking.

I know I am missing this weeks macro-post, but here is something to tide you over

I sat down with Adam for 10 mins yesterday, and we looked at the code example I had made for modifying an audio loop.

It turns out that the code was making a mono edit of the stereo signal. This has now been rectified, and I just need to modify the normalisation function to work with the stereo signal. That should not take too long.

Finally, I will look at modifying the program to change the normalisation offset level, the incured % values to THD per non-linear device. Once this is up and running, I will make a version for each nonlinear equation in the Hawksford & Woon-Seng Gan paper 'Perceptually-Motivated Objective Grading of Nonlinear Processing in Virtual-Bass Systems.

I am also on the hunt for more non-linear equations that describe behaviors of nonlinear devices, particularly loudspeakers and amplifiers. One of the problems with the Hawksford paper normalised models is the normalized parts rely on the exponential function a lot, and are therefor not easy to attenuate the curves of.

More to come shortly.

This week I am also going to look more into literature relating to perceptual models for amounts of distortion and loudness. I am also going to start thinking more about the bits of human hearing that come into effect when thinking about loudness.

I will start by thinking loud, with the Acoustic Reflex.

Miniport 2 of the week - Macropost to come tomorrow hopefully

I am currently using the nonlinear devices from the Hawksford paper 'perceptually motivated... (see last post)' to distort audio and look/listen to the effects.

I now need to figure out how to normalize the audio, so I have some way of keeping track on making similar comparisons.

Here is an example of the time/frequency/amplitude plot (spectrogram) of a clean and distorted version on a signal. I can't currently measure the amount of distortion that is in the signal after processing.

Also this is a useful video for those of us who aren't yet very familiar with matlab:

https://www.youtube.com/watch?v=ie7iREcYBPU

I am getting there!

Micropost 1 of the week

Here are links to a folder containing the IEP proposal and Matlab program.



Props to Adam Hill for his supervision!

And now... Eternity.

Not a comment on infinite regression, though that seem to describe the pattern of the quality of my humour. I am getting pretty frustrated with measuring the THD of music, as opposed to sine waves.

This week so far has been busy, much like any other week but more so. 

I came strait out of a trip to north Wales, into a frenzy of work on my proposal followed by two days of reproduced sound. Yesterday I handed in the proposal. 

I will make it available to you all, and discuss some of the points I found while writing it, in this blog.

Although I said what I needed to say in it (or at least I think so), I feel the proposal was possibly too concise. The volume of references was roughly a similar size to the problem definition and aims.

I had actually finished the proposal about a week before submission, but as always not to a standard I thought was good enough. I thought it would pass, but it wasn't really the standard I thought represented what I wanted to do. My only fear is that the current version is too concise as I have said. We shall see. 

So I refined my search terms/methods and found a vast array of research, into the perception of audio quality and its relationship with distortion. There are already many models around for measuring and predicting distortion, this study may be able to go to a depth I didn't consider before hand.

A particularly good paper is 'Detection of nonlinear distortion in audio signals' by S Mare, which has provided a great look into a method of detecting distortion without needing the original signal for comparison. The really useful thing is the definition of terms and equations available. As it turns out, the IEEE seem to have plenty of research papers which look particularly deeply into THD in reference to power systems, which may undoubtedly come into use at some point. I am starting to think that THD really isn't the be all and end all of distortion measures though.

My THD calculation software is now fully working for the pure sine waves, the problem was in the array addressing.

I am currently looking for papers which focus on nonlinear polynomial distortions, to try and model some realistic system behaviour. It is all go.

It is also interesting to see how many papers which describe THD and methods for measuring related audio quality, nod towards perceived loudness without going into real detail. Voishvillo's 2006 paper 'Assessment of Nonlinearity in Transducers and Sound Systems – from THD to Perceptual Models' is a particularly good example of this. 

I do find it particularly interesting that the recent post-grads who presented at reproduced sound this year, seemed to focus particularly on a mathematical model and not on the aim of using the model.

Foot note:

One of the studies presented at reproduced sound was looking at the quality of internet audio, and how people rate it. As we sat in the lecture, various samples were played. Given the videos were not normalised, I felt that the more distorted audio signals presented sounded louder but not in a good pleasing way, in the instances that they were at the same level i.e. increased level w/m^2 and loudness is not the same thing when taking spectral distribution into account.

Maybe my listening test could be an online one. I would get a more diluted data set, but the increase in participant numbers may help statistically.

It is absolutely time for another blog post!

Last week, I failed to deliver a solid post discussing what has been going on: so here it is.

Duh Duh, MATLAB!

Last time I sat down with Adam & we had a quick chat about my plan.
We then sat down at a laptop with Matlab, and wrote down a program which;

• Generated a sine wave at 1kHz
• Transformed the sine wave using the FFT function, and graphed in the frequency domain 
• Imposed an amplitude limit on the signal
• Graphed the resulting signal in the frequency domain, displaying all extra harmonic content

This weeks plan is to finish the code that measures the % THD, and to test the code. 

Then I will make a 1s loop of the signal at 2.5,5,7.5,10 and 15% and do a quick listening test.
Following that I will run audio through the same system and the same %s and do another quick dirty listening test to see if I am on the right track.

But I have not been totally slacking.

I have spent a reasonable amount of time in the last week finishing off the proposal paperwork, which I will subsequently hand in later today. 

It turns out there is plenty of surrounding research that I have found, but little to nothing on the perception of loudness directly. That could just be my poor Google skills. There is plenty on quality and thresholds (which I suppose are technically loudness), but none on the exact change of perception.

I will be at the Reproduced Sound conference on Tuesday & Wednesday next week, listening to talks such as Leo Beranek's key notes on concert hall design. This may be the last opportunity for me to listen to him lecture. It is also the IOA 40th anniversary, and I may even get the chance to bump into Sam Wise which would be pleasant. 

Back to work following a frozen weekend in North Wales

Will report back sooner, rather than later!

Just a quick update before the main one of the week

Over the past few days I have recieved a few emails from John, which mention some really useful points related to elements of the project.

On Thursday, Adam took me through a little bit of Matlab, and helped me translate into code the signal analysis methods I needed. The program is a basic  FFT and THD analysis tool.

I am having some trouble getting the correct THD numbers however, so more work needs to be done. I also need to make the model analyse for different distortion methods, different levels and different frequencies eventually.

I will spend my spare time over the next few days getting my project proposal finalized.
I am also pouring over THD calculation papers, to try and figure out why the numbers aren't adding up.

Our first meeting with John 29/9/14

I will tidy up the blog format soon.

The Meeting

Fresh back from an eye-opening weekend of climbing at Malham Cove, this morning I jumped strait into a meeting with Anro and  John  from d&b audiotechnik.

After the discussion with John, it seems his perception of the projects are far more integrated than I first thought which is a beautiful thing. Although initializing the skype call was difficult at first, it eventually worked and we got strait to business. The meeting was split into two focused sections with a slight push towards the overlapping elements of the two projects.

The notes i took down (which need to be further expanded upon) with some modification

Anros Parts:

• Dynamics in the perception of loudness
• listener fatigue from low crest factor music (over compression?)
• 12dB peak to rms on modern recordings 
• 4-5dB crest factor/dynamic range on radio
• loudness wars causing listener fatigue 
• tv advertising using the same method to make their adverts much louder
• cut high/low in dance music breakdowns, the drop causing a perceived increase in loudness - this happens in many other genres too - think Pixies - where is my mind?
• ppm6 (bbs design) - brick wall limiters on transmitter for digital output - ppm6 is peak - 8dB above 0dBu - slower decay to allow read of true peaks 
• find the really good book on level metering!!!!
• V U Meters - attempt to show perceived level - analogue components go into saturation
• Florian Camerer - a understanding of loudness - broadcast research
• Broadcasters set off a war on over-compression and increasing signal
• Find the British standard for level monitoring systems - see the attached effects on short & long term
• Florians algorithms for measurement of perceived loudness
• Application from broadcast research - application in the live sound arena at a substantial increase in level
• RESOLUTION MAGAZINE - LINEUP MAGAZINE - some history - institute of broadcast sounds
• Resolution is the upmarket studio magazine - like SOS for pros - find the lineup section - loads of info on loudness monitoring is there
• Software loudness measurement - will (hopefully) make over-loud over-compressed music defunct
• Look for broadcast sound lectures
• The effects of psychoacoustics - (have a chat with Peter Lennox intermittently)
• People expect clubs and gigs to be louder than at home/in headphones
• does the perception of the test change at different levels?
• the limiter and  attack/release time of the ear - find papers WE NEED THIS!
• Twice as loud is 10dB perceptually - the research is out there. Get it!
• 3dB is roughly twice the wattage in system terms - normally 10dB is 10 times the wattage - MONEY!
• 95dB max is normally fine for a generic theater - RnR is about 105dB - the ears protection mechanism is in effect at that level for sure
•  area under curve i.e. long-term crest factor in signals
• military research into the biology of the ears protective mechanism - and the effect of crest factor on the listening protection system - Maybe contact the RAF acoustics corps and see if they can point me towards some research

Simon Parts:

• Where the idea came from - Johns time in the uk techsupport office 
• application support - eas - club installs - DJ redline 
• jbl eons distorting to hell was "loud according to club owner - environmental differences in venue size and acoustic shape of room may have also had a factor when considering increased harmonic content??? Maybe a little bit of pride on the side of the southampton bar owner?
  JBL eons aren't exactly heavy hitters compared to DB C systems... 
• 10% harmonic distortion is the point where djz and club goers through it was loud enough? Maybe? This is according to a person John spoke to... maybe the research is out there...
• adding harmonic distortion, does it work across genres? is it just DJs? in our experience..
  that said, the FFAF engineer and some others were using techniques john mentioned earlier, to do with analogue saturation running the preamps hot and the fader low for increased details
• Are the effects of added harmonics purely psychoacoustics? 
• Crest factor (again)
  
• add harmonic distortion for improved bass?
• get example levels of harmonic distortion - see what people think - a quick and dirty listening test
• see if there is a basis for study on this topic
• Check types of distortion and patterns of harmonics... maybe the measurement of change of patterns will mean a randomized pattern choice with a smear time will be the most suitable option 
• check out inter-mod distortion too
• sort out sample and send to john

book the next meeting (done)

So my ongoing action plan for the next month

• Post up useful related literature 
• Finish proposal
• Find accurate way of measuring THD% in a system
• Plum music through that system (mat lab aplet?)
• Get samples to listen to at different 
• Go and get 30 samples of data - question based? Web Based? Headphones.
• Experiment with different types of distortion
• Experiment with level differences
• Explore crest factor more deeply

It begins today

One useful paper may be "On the Definition of Total Harmonic Distortion and Its Effect on Measurement Interpretation" by Doron Shmilovitz - although this is not audio-specific

Here is a video by Bob Katz on the loudness war:

MY IEP FIRST POST

25/9/14

The Project

Over the summer months I have been researching towards my Independent Engineering Project, which is the 'dissertation' for my degree 'BSc (Hons) Sound, light & Live Event Technology at the University of Derby. 

Along with this module, I have taken on 'Embedded Systems', 'Audio Digital Signal Processing', 'Live Event Practice' and 'Electro-acoustics & Lighting Design'. This should prove to be a monster year!

I was contacted a few weeks ago by a lecturer, who has devised a number of possible IEP projects which will involve being mentored by John Taylor from D&B Audiotechnik. This large and small format loudspeaker and peripheral manufacturer is not only famous for incredible bottom end, but its use of DSP which is something I am very interested in.

After bidding for the project, I was offered the chance to take it on along with another student. After some discussion and the toss of a coin, we chose the elements of the project we wish to work on.

The Background

So the story goes that Mr Taylor goes on occasion to a club where D&BA have installed a sound system. When he visits, the system is often in a state of saturation and high THD as it is pushed too hard, When he fixes the system, the club and clients complain that it is too quiet. As we know, an amplifier can only push out so much level, and no more. When pushed harder, the amplifier will distort the signal and may not get physically louder. As such the question arises, does increased THD make sound subjectively louder?

My initial approach to this is that in principal, yes. Consider energy output as a graphical curve:

• An amplifier may only attenuate so much power (W) 
• If you push an amplifier to produce more power that it can handle it saturates, thus introducing more harmonics (distortion). Thus the waveform is changed. 
• In the instance that this distortion is constructive (considering interference), more harmonics means more energy dissipated under the curve across the frequency domain. 
• An increase in spectral content, even at the same level may excite more of the basilar membrane and may thus be perceived as louder due to a increased energy transfer. 

Thus my IEP will be test research, implement and test methods of adding THD into signals at a constant peak and average level, to discern objective and perceptive loudness (increase or decrease) with different levels and styles of THD added. 

The Major Questions

• Are we so used to distortion that we expect it? Do we need it?
• Is there a link between THD and the increase in feeling the music (getting punched with bass)
• Could the introduction of THD be beneficial to audio-reproduction both in the isolated and free listening formats (headphones and in the club/field)
• Could adding the correct kind of distortion save loudspeaker systems?
• Does increased THD actually increase the energy output by a loudspeaker/system?

I am now beginning my project proposal.