Tag Archives: convolution

The Different “Sides” of Convolution

We all know how much convolution is used in DSP, and what a significant part it has in making many effects and analysis techniques a reality.  Often we hear about FFT, or fast convolution, but in actuality these algorithms are only faster than straight convolution with a kernel length (of the impulse response) greater than around 60 – 64 or so.  Anything shorter than that is best handled with normal convolution, which can be implemented using either an input side or output side algorithm.

The input side algorithm is normally the one that we learn first (at least it was for me, and judging from many texts and books on it, it seems to be common).  However, the output side algorithm is perhaps slightly easier to code, and we’ll see why later on.  For this post I’m going to evaluate the effects each algorithm has on the resulting audio signal because each one could have an advantage in some situations over the other.  I’ll also share a little trick that will speed up the convolution process for certain types of filters.

The output side algorithm performs the convolution, as one might expect, from the viewpoint of the output (or the result) of the operation, while the input side algorithm implements convolution from the viewpoint of the input signal.  To look at it another way, recall that convolving an input signal with an impulse response results in an output signal that is the length of the input + the length of the impulse response – 1.  With the input side algorithm, this “extra” bit of length is appended to the resulting signal, and is probably the easiest one to theoretically grasp.  The output side algorithm looks at what points from the input we need in order to get our resulting signal.

The reason why the output side method feels more natural to code is that we often write in terms of the output or result: y(n) = ____ — the output is equal to some kind of expression, for example.  Approaching convolution this way requires input samples on the negative side (x[-2], x[-1], etc.), which of course don’t exist.  To get around that little problem, one common way is to implement a delay line that contains all zeroes at first, then shifting in the input signal as we go, doing the calculating with the impulse response and the delay line instead of direclty with the input.

I’ve been busy redesigning and overhauling my DSP filter class, which has entailed some experimentation with these two convolution methods on FIR filters.  Here we can see visually the difference between them.  First up is the output side convolution of a windowed-sinc FIR filter using a Kaiser window of order 162 on a simple triangle wave (using test oscillators is a good way to visually see what’s happening).

Output side convolution on a triangle wave.

Output side convolution on a triangle wave.

I purposely made the filter kernel (impulse response) fairly long so it would illustrate the delay and shift that’s happening at the start of the signal due to the output side algorithm.  Compare that with the image of the input side algorithm using exactly the same filter and order.

Input side convolution on a triangle wave.

Input side convolution on a triangle wave.

Huge difference.  It’s clearly audible as well in this test signal (samples below).  With the output side algorithm there is a clear popping sound as the audio abruptly starts in contrast to the much smoother beginning of the test signal filtered with the input side algorithm.

Here you can hear the difference (to hear it clearly you may have to download the files and listen in an audio editor to avoid the glitchy start of web browser plug-ins):

Triangle wave filtered with output side

Triangle wave filtered with input side

Now to return to the point about the output side method being perhaps a little more natural to code.  The little wrinkle we face in implementing the input side algorithm is that with block processing (which is so common in DSP), how do we account for the longer output as a result of the convolution?  In many cases we don’t have control over the size of the audio buffers we’re given or have to work with.  The solution is to use the overlap-add method.

It’s really not more difficult than the output side algorithm.  All we have to do is calculate the convolution fully (into our own internal buffer if we have to), save the extra length at the end (which will be equal to the impulse response length), then add that to the start of the next block.  Rinse and repeat.  Here is C++ code that implements both of these convolution methods.

Output side convolution process.

Output side convolution process.

Input side convolution process.

Input side convolution process.

Don’t worry, I didn’t forget to share the trick I mentioned at the beginning.  Many FIR filters have linear phase response, which means that their kernels are symmetrical.  That provides us a great opportunity to eliminate extra calculations that aren’t needed.  So notice in the above code each ‘j’ loop (the kernel or impulse response loop) only traverses half the kernel length, as the value of input[i] * mKernel[j] is the same as input[i] * mKernel[mKernelLength-j-1].  At the end of the kernel’s loop we calculate the midpoint value.

Again, the calculations involving symmetry is perhaps easier to see in the output side algorithm, because we naturally gravitate towards expressions that result in a single output value.  If you work a small convolution problem out on paper, however, it will help you see the input side algorithm and why it works.

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Audio Resampling: Part 1

Resampling in digital audio has two main uses.  One is to convert audio into a sampling rate needed by a particular system or engine (e.g. converting 48kHz audio to the required 44.1kHz required by CDs).  The second is to avoid aliasing during signal processing by raising the Nyquist limit.  I will be discussing the latter.

Lately I’ve been very busy working on improving and enhancing the sound of ring modulation for a fairly basic plug-in being developed by AlgoRhythm Audio (coming soon).  I say basic becuase as far as ring modulation goes, there are few DSP effects that are simpler in theory and in execution.  Simply take some input signal, multiply it by a carrier signal (usually some kind of oscillator like a sine wave), and we have ring modulation.  The problem that arises, however, and how this connects in with resampling, is that this creates new frequencies in the resulting output that were not present in either signal prior to processing.  These new frequencies created could very well violate the Nyquist limit of the current sampling rate during processing, and that leads us to resampling as a way to clamp down on aliasing frequencies that can be introduced as a result.

Aliasing is an interesting phenomenon that occurs in digital audio, and in every sense of the word is an undesired noise that we need to make sure does not pollute our audio.  There are many resources around that go into more detail on aliasing, but I will give a brief overview of it with some audio and visual samples.

Aliasing occurs when there are frequencies present in a signal that are greater than the Nyquist limit (half of the sampling rate).  What happens in such a case is that the sampling rate is not high enough to properly capture (sample) the high frequency of the signal, and so the frequency “folds over” and creates aliases that are mirror images of the original frequency.  Here, for example, is a square wave at 4000Hz created using 20 harmonics at a sampling rate of 44.1kHz (keep in mind that 4000Hz is the fundamental frequency, and that square waves contain many additional frequencies above that depending on how many harmonics were used to create it, so in this case Nyquist is still being violated):

4000Hz square wave with aliasing

Notice the low tone below the actual 4000Hz frequency.  Here is the resulting waveform of this square wave that shows us visually that we’re not sampling fast enough to accurately reproduce the waveform.  Notice the inconsistencies in the waveform.

squarewave_alias

Now, going to the extreme a bit, here is the same square wave sampled at a rate of 192kHz.

4000Hz square wave with no aliasing

It’s a pure 4000Hz square wave tone.  Examining the waveform of this square wave shows us that the sampling rate was more that adequate to reproduce this signal digitally:

square_noalias

Not all signal processing effects are susceptible to aliasing, and certainly not to the same degree.  Because ring modulation produces additional inharmonic frequencies, it is a prime example of a process that is easily affected by aliasing.  Others include distortion and various other kinds of modulation techniques (especially when taken to the extreme).  However, ring modulation with a sine wave is generally safe as long as the frequency of the sine wave is kept  low enough because sine waves have no harmonics to them, only the fundamental.  Introducing other wavetypes into this process, however, can quite easily bring about aliasing.

Here is an example of ring modulation with a sawtooth wave sweeping up from about 200Hz to 5000Hz.  As it glides up, you will be able to hear the aliasing kicking in at around 0:16 or so.  The example with no aliasing has been upsampled by 3X before processing, then downsampled by 3X back to its original rate.

Ring modulation with a sawtooth sweep, with aliasing

Ring modulation with sawtooth sweep, with no aliasing

So how does upsampling and downsampling work?  In theory, and even to some extent in practice, it’s very straightforward.  The issue, as we will see, is in making it efficient and fast.  In DSP we’re always concerning ourselves with speedy execution times to avoid latency or audio dropouts or running out of memory, etc.

To upsample, we insert 0-valued samples in between every L-th (our upsampling factor) original sample.  (i.e. upsampling by 3X, [1, 2, 3, 4] becomes [1,0,0,2,0,0,3,0,0…])  However, this introduces aliasing into our audio so we need to interpolate these values so that they “join” the sample values of the original waveform.  This is accomplished by using an interpolating low-pass filter.  This entire process is known as interpolation.

Dowsampling is much the same.  We remove every M-th (our downsampling factor) sample from the original signal.  (i.e. downsampling by 2X, [1,2,3,4,5] becomes [1,3,5…])  This process does not introduce aliasing, but we do need to make sure the Nyquist limit is adhered to at the new sampling rate by low-pass filtering with a cutoff frequency at the new Nyquist rate.  This entire process is known as decimation.

Fairly straightforward.  Proceeding with this method, however, would be known as brute force — generally not a good way to go.  The reason why becomes clearer when we consider what kind of low-pass filter we need for this operation.  The ideal filter would be one that would brick-wall attenuate all frequencies higher than Nyquist and leave everything else untouched (thus preserving all the frequencies and tonal content of our original audio).  This is, alas, impossible, as it would require an infinitely long filter kernel.  The function that would implement this ideal filter is the rectangular function.

rect

The rectangular function.

By taking the Fourier transform of the rectangular function we end up with the sinc function, which is given by:

y(x) = sin (x) / x, which becomes y(x) = sin (πx) / πx for signal processing.

Graph of the sinc function, sin (x) / x

Graph of the sinc function, sin (πx) / πx

The sinc functions trails on for infinity in both directions, which can be seen in the graph above, so we need to enforce bounds around it by applying a window function.  Windowing is a method of designing FIR filters by essentially “surrounding” a function (in our case the sinc) by the window, which enforces bounds so that we can properly derive a filter kernel for use in calculations.  The rectangular function shown above is a type of window, but as I mentioned, infinite slope is a deal-breaker in audio.

The Blackman window, given by the function

w(i) = 0.42 – 0.5 cos(2πi M) + 0.08 cos(4πi M),

where M is the length of the filter kernel, is a good choice for resampling because it offers a good stop-band attenuation of -74dB with good rolloff.  Putting this together with the sinc function, we can derive the filter kernel with the following formula:

Windowed-sinc kernel formula*

Windowed-sinc kernel formula*

where fc is the normalized cutoff frequency.  When i = M/2, to avoid a divide by zero, h(i) = 2fc.  K is a constant value used to achieve unity gain at zero frequency and can be ignored while calculating the kernel coefficients.  After all coefficients have been calculated, K can be found by summing together all the coefficients, h(i), and then dividing each by the resulting sum.

* Source: Smith, Steven W., “The Scientist and Engineer’s Guide to Digital Signal Processing”, Chapter 16.

Now that we have the filter design, let’s consider the properties of the FIR filter and compare them to IIR filter designs.  IIR filters give us better performance and attenuation at lower orders, meaning that they execute faster and perform better with fewer calculations than FIR filters.  However, IIR filters are still not powerful enough, even at slightly higher orders, to give us the performance we need for resampling, and trying to push IIR filters into very high orders can make them unstable and/or susceptible to quantization error due to the nature of recursion.  IIR filters also do not offer linear phase response.  FIR filters are the better choice for these reasons, but the unfortunate drawback is that they execute slowly due to being implemented by convolution.  In addition, they need to be pushed to high orders to give us the performance needed for attenuating aliasing frequencies.

However, the order of the interpolating low-pass filter can be negotiated based on the frequency content of the audio signal(s) involved.  If the audio is sufficiently oversampled, it will not contain enough frequencies near Nyquist, and as such a lower order filter can be used with a gentler rolloff without adversely affecting the audio and attenuating actual frequencies in the signal.  There are plenty of cases where we just don’t know the frequency content of the audio signals involved in processing, however, so a strong filter may be needed in these cases.  Here is a graph of a 264-order window-sinc filter (in other words, a filter kernel of length 265 including the sample x(0)):

264-order window-sinc low-pass filter frequency response

264-order window-sinc low-pass filter frequency response (cutoff frequency at 10kHz, resulting in a transition band of 882Hz)

With this in consideration, it can be easy to see that convolving a signal with a 264-order FIR filter is computationally costly for real-time processing.  There are a number of ways to improve upon this when it comes to resampling.  One is using the FFT to apply the filter.  Another interesting solution is to combine the upsampling/downsampling process into the filter itself, which can further be optimized by turning it into a polyphase filter.

The theory and implementation of a polyphase filter is a fairly long and involved topic on its own so that will be forthcoming in part 2,where we look at how to implement resampling efficiently.