Hypothetically, how an FFT-based equalizer can be programmed.

One of the concepts which I only recently posted about was, that I had activated an equalizer function, that was once integrated into how the PulseAudio sound server works, but which may be installed with additional packages, in more-recent versions of Debian Linux. As I wrote, to activate this under Debian 8 / Jessie was a bit problematic at first, but could ultimately be accomplished. The following is what the controls of this equalizer look like, on the screen:

Equalizer_1

And, this is what the newly-created ‘sink’ is named, within the (old) KDE-4 desktop manager’s Settings Panel:

Equalizer_2

What struck me as remarkable about this, was its naming, as an “FFT-based Equalizer…”. I had written an earlier posting, about How the Fast Fourier Transform differs from the Discrete Fourier Transform. And, because I tend to think first, about how convolutions may be computed, using a Discrete Cosine Transform, it took me a bit of thought, to comprehend, how an equalizer function could be implemented, based on the FFT.

BTW, That earlier posting which I linked to above, has as a major flaw, a guess on my part about how MP3 sound compression works, that makes a false assumption. I have made more recent postings on how sound-compression schemes work, which no longer make the same false assumption. But otherwise, that old posting still explains, what the difference between the FFT and other, Discrete Transforms is.

So, the question which may go through some readers’ minds, like mine, would be, how a graphic equalizer based on the FFT can be made computationally efficient, to the maximum. Obviously, when the FFT is only being used to analyze a sampling interval, what results is a (small) number of frequency coefficients, spaced approximately uniformly, over a series of octaves. Apparently, such a set of coefficients-as-output, needs to be replaced by one stream each, that isolates one frequency-component. Each stream then needs to be multiplied by an equalizer setting, before being mixed into the combined equalizer output.

I think that one way to compute that would be, to replace the ‘folding’ operation normally used in the Fourier Analysis, with a procedure, that only computes one or more product-sums, of the input signal with reference sine-waves, but in each case except for the lowest frequency, over only a small fraction of the entire buffer, which becomes shorter according to powers of 2.

Thus, it should remain constant that, in order for the equalizer to be able to isolate the frequency of ~31Hz, a sine-product with a buffer of 1408 samples needs to be computed, once per input sample. But beyond that, determining the ~63Hz frequency-component, really only requires that the sine-product be computed, with the most recent 704 samples of the same buffer. Frequency-components that belong to even-higher octaves can all be computed, as per-input-sample sine-products, with the most-recent 352 input-samples, etc. (for multiples of ~125Hz). Eventually, as the frequency-components start to become odd products of an octave, an interval of 176 input samples can be used, for the frequency-components belonging to the same octave, thus yielding the ~500Hz and ~750Hz components… After that, in order to filter out the ~1kHz and the ~1.5kHz components, a section of the buffer only 88 samples long can be used…

Mind you, one alternative to doing all that would be, to apply a convolution of fixed length to the input stream constantly, but to recompute that convolution, by first interpolating frequency-coefficients between the GUI’s slider-positions, and then applying one of the Discrete Cosine Transforms to the resulting set of coefficients. The advantage to using a DCT in this way would be, that the coefficients would only need to be recomputed once, every time the user changes the slider-positions. But then, to name the resulting equalizer an ‘FFT-based’ equalizer, would actually be false.

(Updated 6/23/2020, 18h50… )

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Getting the integrated equalizer to work, from Debian Jessie, KDE 4.

I happen to have an older laptop, which I name ‘Klystron’, that is running Debian 8 / Jessie, with KDE 4 as its desktop manager. Don’t ask me why, but I tend to leave older builds of Linux running on some of my computers, just because they seem to be running without any deficiencies.

That laptop has lousy speakers. I decided a few days ago, that it would benefit, if I could get the 14-band graphical equalizer to work, that is generally available on Linux computers which, like that laptop, use the ‘PulseAudio’ sound server. However, on this old version of Linux, achieving that was a bit harder than it’s supposed to be. Yet, because I succeeded, I felt that I should share with the community, what the steps were, needed to succeed.

First of all, this is what the equalizer looks like, which I can now open on that laptop:

Equalizer_1

And it works!

 

In order to get this sort of equalizer working with PulseAudio, eventually, the following two modules need to be loaded:

module-equalizer-sink

module-dbus-protocol

And, if I gave the command ‘load-module…’ naively from the command-line, as user, because under my builds of Linux, PulseAudio runs in user mode, both these modules seem to load fine, without my having to install any additional packages.

On more recent builds of Linux, one needs to install the package ‘pulseaudio-equalizer’ to obtain this feature, or some similarly-named package. But, because these two modules just loaded fine under Debian / Jessie, I guess the functionality once came integrated with PulseAudio.

But I soon started to run in to trouble with these modules, and discovered why, then, the equalizer function was not set up to run out-of-the-box…

(Updated 6/26/2020, 10h30… )

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Some realizations about Digital Signal Processing

One of the realizations which I’ve just come across recently, about digital signal processing, is that apparently, when up-sampling a digital stream twofold, just for the purpose of playing it back, simply to perform a linear interpolation, to turn a 44.1kHz stream into an 88.2kHz, or a 48kHz stream into a 96kHz, does less damage to the sound quality, than I had previously thought. And one reason I think this is the factual realization that to do so, really achieves the same thing that applying a (low-pass) Haar Wavelet would achieve, after each original sample had been doubled. After all, I had already said, that Humans would have a hard time being able to hear that this has been done.

But then, given such an assumption, I think I’ve also come into more realizations, of where I was having trouble understanding what exactly Digital Signal Processors do. It might be Mathematically true to say, that a convolution can be applied to a stream after it has been up-sampled, but, depending on how many elements the convolution is supposed to have, whether or not a single DSP chip is supposed to decode both stereo channels or only one, and whether that DSP chip is also supposed to perform other steps associated with playing back the audio, such as, to decode whatever compression Bluetooth 4 or Bluetooth 5 have put on the stream, it may turn out that realistic Digital Signal Processing chips just don’t have enough MIPS – Millions of Instructions Per Second – to do all that.

Now, I do know that DSP chips exist that have more MIPS, but then those chips may also measure 2cm x 2cm, and may require much of the circuit-board they are to be soldered in to. Those types of chips are unlikely to be built-in to a mid-price-range set of (Stereo) Bluetooth Headphones, that have an equalization function.

But what I can then speculate further is that some combination of alterations of these ideas should work.

For example, the convolution that is to be computed could be computed on the stream before it has been up-sampled, and it could then be up-sampled ‘cheaply’, using the linear interpolation. The way I had it before, the half-used virtual equalizer bands would also accomplish a kind of brick-wall filter, whereas, to perform the virtual equalizer function on the stream before up-sampling would make use of almost all the bands, and doing it that way would halve the amount of MIPS that a DSP chip needs to possess. Doing it that way would also halve the frequency linearly separating the bands, which would have created issues at the low end of the audible spectrum.

Alternatively, implementing a digital 9- or 10-band equalizer, with the
bands spaced an octave apart, could be achieved after up-sampling, instead of before up-sampling, but again, much more cheaply in terms of computational power required.

Dirk

LG Tone Infinim HBS-910 Bluetooth Headphones

In This earlier posting, I had written that my LG Tonepro HBS-750 Bluetooth Headphones had permanently failed. Today, I received the HBS-910 headphones that are meant to replace those. And as I’ve written before, it is important to me, to benefit from the high-quality sound, that both sets of headphones offer.

I’m breaking in the new ones, as I’m writing this.

There exists a design-philosophy today, according to which music-playback is supposed to boost the bass and attenuate the highest frequencies – the ones higher than 10kHz – so that the listener will get the subjective impression that the sound is ‘louder’, and so that the listener will reduce the actual signal-level, to preserve their hearing better than it was done a few decades ago.

  1. The lowest-frequency (default) setting on the equalizer of the headphones does both of those things.
  2. The next setting stops boosting the bass.
  3. The third setting, stops attenuating the treble.

Overall, I get the impression that the highest frequencies which the HBS-910 can reproduce, extend higher, than what the HBS-750 was able to reproduce.

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