Room correction. What are we trying to achieve?

The short version…

The recent availability of DSP is leading some people to assume that speakers are, and have always been, ‘wrong’ unless EQ’ed to invert the room’s acoustics.

In fact, our audio ancestors didn’t get it wrong. Only a neutral speaker is ‘right’, and the acoustics of an average room are an enhancement to the sound. If we don’t like the sound of the room, we must change the room – not the sound from the speaker.

DSP gives us the tools to build a more neutral speaker than ever before.


There are endless discussions about room correction, and many different commercial products and methods. Some people seem to like certain results while others find them a little strange-sounding.

I am not actually sure what it is that people are trying to achieve. I can’t help but think that if someone feels the need for room correction, they have yet to hear a system that sounds so good that they wouldn’t dream of messing it up with another layer of their own ‘EQ’.

Another possibility is that they are making an unwarranted assumption based on the fact that there are large objective differences between the recorded waveform and what reaches the listener’s ears in a real room. That must mean that no matter how good it sounds, there’s an error. It could sound even better, right?

No.

A reviewer of the Kii Three found that that particularly neutral speaker sounded perfect straight out of the box.

“…the traditional kind of subjective analysis we speaker reviewers default to — describing the tonal balance and making a judgement about the competence of a monitor’s basic frequency response — is somehow rendered a little pointless with the Kii Three. It sounds so transparent and creates such fundamentally believable audio that thoughts of ‘dull’ or ‘bright’ seem somehow superfluous.”

The Kii Three does, however, offer a number of preset “contour” EQ options. As I shall describe later, I think that a variation on this is all that is required to refine the sound of any well-designed neutral speaker in most rooms.

A distinction is often made between correction of the bass and higher frequencies. If the room is large, and furnished copiously, there may be no problem to solve in either case, and this is the ideal situation. But some bass manipulation may be needed in many rooms. At a minimum, the person with sealed woofers needs the roll-off at the bottom end to start at about the right frequency for the room. This, in itself, is a form of ‘room correction’.

The controversial aspect is the question of whether we need ‘correction’ higher up. Should it be applied routinely (some people think so), as sparingly as possible, or not at all? And if people do hear an improvement, is that because the system is inherently correcting less-than-ideal speakers rather than the room?

Here are some ways of looking at the issue.

  1. Single room reflections give us echoes, while multiple reflections (of reflections) give us reverberation. Performing a frequency response measurement with a neutral transducer and analysing the result may show a non-flat FR at the listening position even when smoothed fairly heavily. This is just an aspect of statistics, and of the geometry and absorptivity of the various surfaces in the room. Some reflections will result in some frequencies summing in phase, to some extent, and others not.
  2. Experience tells us that we “hear through” the room to any acoustic source. Our hearing appears not to be just a frequency response analyser, but can separate direct sound from reflections. This is not a fanciful idea: adaptive software can learn to do the same thing.

The idea is also supported by some of the great and the good in audio.

Floyd Toole:

“…we humans manage to compensate for many of the temporal and timbral variations contributed by rooms and hear “through” them to appreciate certain essential qualities of sound sources within these spaces.”

Or Meridian’s Bob Stuart:

“Our brains are able to separate direct sound from the reverberation…”

  1. If we EQ the FR of the speaker to obtain a flat in-room measured response including the reflections in the measurement, it seems that we will subsequently “hear through” the reflections to a strangely-EQ’ed direct sound. It will, nevertheless measure ‘perfectly’.
  2. Audio orthodoxy maintains that humans are supremely insensitive to phase distortion, and this is often compounded with the argument that room reflections completely swamp phase information so it is not worth worrying about. This denies the possibility that we “hear through” the room. Listening tests in the past that purportedly demonstrated our inability to hear the effects of phase have often been based on mono only, and didn’t compare distorted with undistorted phase examples – merely distorted versus differently distorted, played on the then available equipment.
  3. Contradicting (4), audiophiles traditionally fear crossovers because the phase shifts inherent in (non-DSP) crossovers are, they say, always audible. DSP, on the other hand, allows us to create crossovers without any phase shift i.e. they are ‘transparent’.
  4. At a minimum, speaker drivers on their baffles should not ‘fight’ each other through the crossover – their phases should be aligned. The appropriate delays then ensure that they are not ‘fighting’ at the listener’s position. The next level in performance is to ensure that their phases are flat at all frequencies i.e. linear phase. The result of this is the recorded waveform preserved in both frequency and time.
  5. Intuitively, genuine stereo imaging is likely to be a function of phase and timing. Preserving that phase and timing should probably be something we logically try to do. We could ‘second guess’ how it works using traditional rules of thumb, deciding not to preserve the phase and timing, but if it is effectively cost-free to do it, why not do it anyway?
  6. A ‘perfect’ response from many speaker/room combinations can be guaranteed using DSP (deconvolution with the impulse response at that point, not just playing with a graphic equaliser). Unfortunately, it will only be valid for a single point in space, and moving 1mm from there will produce errors and unquantifiable sonic effects. Additionally, ‘perfect’ refers to the ‘anechoic chamber’ version of the recording, which may not be what most people are trying to achieve even if the measurements they think they seek mean precisely that.
  7. Room effects such as (moderate) reverberation are a major difference between listening with speakers versus headphones, and are actually desirable. ‘Room correction’ would be a bad thing if it literally removed the room from the sound. If that is the case, what exactly do we think ‘room correction’ is for?
  8. Even if the drivers are neutral (in an anechoic situation) and crossed over perfectly on axis, they are of finite size and mounted in a box or on a baffle that has a physical size and shape. This produces certain frequency-dependent dispersion characteristics which give different measured, and subjective, results in different rooms. Some questions are:
    • is this dispersion characteristic a ‘room effect’ or a ‘speaker effect’. Or both?
    • is there a simple objective measurement that says one result is better than any other?
    • is there just one ‘right’ result and all others are ‘wrong’?
  1. Should room correction attempt to correct the speaker as well? Or should we, in fact, only correct the speaker? Or just the room? If so, how would we separate room from speaker in our measurements? Can they, in fact, be separated?

I think there is a formula that gives good results. It says:

  • Don’t rely on feedback from in-room measurements, but do ‘neutralise’ the speaker at the most elemental levels first. At every stage, go for the most neutral (and locally correctable) option e.g. sealed woofers, DSP-based linear phase crossovers with time alignment delays.
  • Simply avoid configurations that are going to give inherently weird results: two-way speakers, bass reflex, many types of passive crossover etc. These may not even be partially correctable in any meaningful way.
  • Phase and time alignment are sacrosanct. This is the secret ingredient. You can play with minor changes to the ‘tone colour’ separately, but your direct sound must always maintain the recording’s phase and time alignment. This implies that FIR filters must be used, thus allowing frequency response to be modified independently of phase.
  • By all means do all the good stuff regarding speaker placement, room treatments (the room is always ‘valid’), and avoiding objects and asymmetry around the speakers themselves.
  • Notionally, I propose that we wish to correct the speaker not the room. However, we are faced with a room and non-neutral speaker that are intertwined due to the fact that the speaker has multiple drivers of finite size and a physical presence (as opposed to being a point source with uniform directivity at all frequencies). The artefacts resulting from this are room-dependent and can never really be ‘corrected’ unambiguously. Luckily, a smooth EQ curve can make the sound subjectively near enough to transparent. To obtain this curve, predict the baffle step correction for each driver using modelling or standard formula with some some trial-and-error regarding the depth required (4, 5, 6 dB?); this is a very smooth EQ curve. Or, possibly (I haven’t done this myself), make many FR measurements around the listening area, smooth and average them together, and partially invert this, again without altering phase and time alignment.
  • You are hearing the direct sound, plus separately-perceived ‘room ambience’. If you don’t like the sound of the ambience, you must change the room, not the direct sound.

Is there any scientific evidence for these assertions? No more nor less than any other ‘room correction’ technique – just logical deduction based on subjective experience. Really, it is just a case of thinking about what we hear as we move around and between rooms, compared to what the simple in-room FR measurements show. Why do real musicians not need ‘correction’ when they play in different venues? Do we really want ‘headphone sound’ when listening in rooms? (If so, just wear headphones or sit closer to smaller speakers).

This does not say that neutral drivers alone are sufficient to guarantee good sound – I have observed that this is not the case. A simple baffle step correction applied to frequency response (but leaving phase and timing intact) can greatly improve the sound of a real loudspeaker in a room without affecting how sharply-imaged and dynamic it sounds. I surmise that frequency response can be regarded as ‘colour’ (or “chrominance” in old school video speak), independent of the ‘detail’ (or “luminance”) of phase and timing. We can work towards a frequency response that compensates for the combination of room and speaker dispersion effects to give the right subjective ‘colour’ as long as we maintain accurate phase and timing of the direct sound.

We are not (necessarily) trying to flatten the in-room FR as measured at the listener’s position – the EQ we apply is very smooth and shallow – but the result will still be perceived as a flat FR. Many (most?) existing speakers inherently have this EQ built in whether their creators applied it deliberately, or via the ‘voicing’ they did when setting the speaker up for use in an average room.

In conclusion, the summary is this:

  • Humans “hear through” the room to the direct sound; the room is perceived as a separate ‘ambience’. Because of this, ‘no correction’ really is the correct strategy.
  • Simply flattening the FR at the listening position via EQ of the speaker output is likely to result in ‘peculiar’ perceived sound, even if the in-room measurements purport to say otherwise.
  • Speakers have to be as rigorously neutral as possible by design, rather than attempting to correct them by ‘global feedback’ in the room.
  • Final refinement is a speaker/room-dependent, smooth, shallow EQ curve that doesn’t touch phase and timing – only FIR filters can do this.

[Last updated 05/04/17]

Advertisements

Data Over Sound

Just saw this mentioned. It’s interesting how an idea that, years ago, was just a method of harnessing existing technology, can re-appear as something funky and brand new. It joins those other technologies that aim to get data into our devices via cost-free, non-contact interfaces, such as QR Codes.

What is Chirp?

A Chirp™ is a sonic barcode. With Chirp technology, data and content can be encoded into a unique audio stream. Any device with a speaker can transmit a chirp and most devices with a microphone can decode them.

People of a certain age will be familiar with the use of audio cassettes as storage for their microcomputer programs back in the 1980s – I think I used reel-to-reel for a time.

I also remember, round about 1980, sending a computer program over the phone to a friend’s house by holding the phone close to the speaker and picking the sound up at the other end with a microphone. As I recall, our version wasn’t really very reliable or practical, but I think we did succeed in sending a short program. Obviously we were inspired by the audio coupler modems that we might have seen in films and documentaries.

full

SMPTE and MIDI timecodes can be recorded as audio signals on analogue tape and can survive multiple transfers and, I dare say, would be robust enough to work over a speaker-microphone link.

In the 1990s we were all familiar with ‘the sound of data’ when we used dial-up modems.

Over the years we have also had DTMF dialling, audio watermarking, Shazam, Siri, Alexa etc. and phone-based automated systems using speech recognition, all of which have to deal with extracting ‘data’ from noisy audio. You would think that the new audio barcodes should be pretty simple to make work reliably.

The Secret Science of Pop

secret-science-of-pop

In The Secret Science of Pop, evolutionary biologist Professor Armand Leroi tells us that he sees pop music as a direct analogy for natural selection. And he salivates at the prospect of a huge, complete, historical data set that can be analysed in order to test his theories.

He starts off by bringing in experts in data analysis from some prestigious universities, and has them crunch the numbers on the past 50 years of chart music, analysing the audio data for numerous characteristics including “rhythmic intensity” and “agressiveness”. He plots a line on a giant computer monitor showing the rate of musical change based on an aggregate of these values. The line shows that the 60s were a time of revolution – although he claims that the Beatles were pretty average and “sat out” the revolution. Disco, and to a lesser extent punk, made the 70s a time of revolution but the 80s were not.

He is convinced that he is going to be able to use his findings to influence the production of new pop music. The results are not encouraging: no matter how he formulates his data he finds he cannot predict a song’s chart success with much better than random accuracy. The best correlation seems to be that a song’s closeness to a particular period’s “average” predicts high chart success. It is, he says, “statistically significant”.

Armed with this insight he takes on the role of producer and attempts to make a song (a ballad) being recorded at Trevor Horn’s studio as average as possible by, amongst other things, adjusting its tempo and adding some rap. It doesn’t really work, and when he measures the results with his computer, he finds that he has manoeuvred the song away from average with this manual intervention.

He then shifts his attention to trying to find the stars of tomorrow by picking out the most average song from 1200 tracks that have been sent into BBC Radio 1 Introducing. The computer picks out a particular band who seem to have a very danceable track, and in the world’s least scientific experiment ever, he demonstrates that a BBC Radio 1 producer thinks it’s OK, too.

His final conclusion: “We failed spectacularly this time, but I am sure the answer is somewhere in the data if we can just find it”.

My immediate thoughts on this programme:

-An entertaining, interesting programme.

-The rule still holds: science is not valid in the field of aesthetic judgement.

-If your system cannot predict the future stars of the past, it is very unlikely to be able to predict the stars of the future.

-The choice of which aspects of songs to measure is purely subjective, based on the scientist’s own assumptions about what humans like about music. The chances of the scientist not tweaking the algorithms in order to reflect their own intuitions are very remote. To claim that “The computer picked the song with no human intervention” is stretching it! (This applies to any ‘science’ whose main output is based on computer modelling).

-The lure of data is irresistible to scientists but, as anyone who has ever experimented with anything but the simplest, most controlled, pattern recognition will tell you, there is always too much, and at the same time never enough, training data. It slowly dawns on you that although theoretically there may be multidimensional functions that really could spot what you are looking for, you are never going to present the training data in such a way that you find a function with 100%, or at least ‘human’ levels of, reliability.

-Add to that the myriad paradoxes of human consciousness, and of humans modifying their tastes temporarily in response to novelty and fashion – even to the data itself (the charts) – and the reality is that it is a wild goose chase.

(very relevant to a post from a few months ago)