I'm writing a program to emulate a vocoder and it obviously works, but any speech that comes through is very difficult to understand. I'm going to explain exactly how my implementation works and then maybe somebody with experience with vocoders or DSP can help me.

128 samples of the voice signal go through a Fast Fourier Transform every audio frame. I then take the absolute value of the complex outputs to get the formant of the voice. I chose this rather than multiple band pass filters and envelope followers because it's faster just to do an FFT. I am 100% sure that this is all done correctly.

My carrier wave is run through several band pass filters centered around frequencies corresponding to the FFT output frequencies. To construct the output I multiply the output of these filters by the corresponding amplitudes in the voice formant obtained from the FFT and add all of these products.

There's a few things that I suspect may be causing issues. The algorithm I used to implement the band pass filters is probably not the best. I feed the output of a low pass filter into a high pass filter. I got the algorithms for the filters under the "Algorithmic Implementation" section of the low pass and high pass filter articles on Wikipedia. I have a strong feeling that these simple filters may not be attenuating unwanted frequencies very well but I can't find a better way to make my band pass filter. I also suspect that my carrier wave might not be very good. As I understand it, you need a carrier wave rich in lots of different frequency content (so obviously not a sine wave). I've listened to a lot of vocoders and the carrier wave usually sounds like it's got some dissonant notes in it. I've tried additive synthesis and FM synthesis but I just can't get a carrier wave that doesn't sound too "clean".

If somebody could give me a better band pass algorithm (if that's what I need), a good formula for a carrier wave, or point out some other likely problem I would be very grateful.

ALSO, "vocoder" is not an available tag. Maybe somebody with enough permissions could kindly add this tag? Also, there's a tag for signal-processing but not digital-signal-processing or DSP, I'm not sure if it would be appropriate to add one of these tags or not.


1 Answer 1


First, it seems to me that you are doing the analysis incorrectly. 128 linearly spaced channels is both too high (it won't capture a smoothed spectral envelope - vocoders usually have at most ~30 channels) and too small (it has a very poor resolution in the lowest frequencies and won't discriminate different formants). You really need to replace that by a bank of constant-Q band-pass filters, 2 or 3 bands / octave to get something that is coarse enough to capture the spectral envelope without picking individual frequency peaks, yet discriminates the frequencies of interest.

Your analysis and synthesis filters can be implemented with biquads. Running a dozen of those in parallel is not cost-prohibitive compared to a FFT.

As for the carrier wave, you can start with a sawtooth or narrow pulse - avoid waveforms like triangle, square, or sine whose spectrum have "gaps". Then stack a few of them to play a chord. You can thicken the carrier by adding slightly detuned version of each component or octaves. Note that in order to achieve the best sounding results, band-limited synthesis (for example with minBLEPs) has to be used to synthesize the square or sawtooth waves.

  • $\begingroup$ Okay, I think I understand the biquad filters. But looking on Wikipedia at the Direct Form 1 (en.wikipedia.org/wiki/Digital_biquad_filter#Direct_Form_1), can't coefficients be combined for better efficiency? For example, the terms (b1)x[n-1] and -(a1)x[n-1] could be combined to have a single coefficient (we'll call (c1)) equal to (b1)-(a1), then those terms could become (c1)x[n-1]. This is correct, is it not? Or am I missing something? Also, would I get significantly better sound using band-limited synthesis? And what should I use for an envelope follower? $\endgroup$
    – Void Star
    Commented Jul 27, 2012 at 20:17
  • $\begingroup$ @BigEndian: as said in that very Wikipedia article, you can improve the efficiency; you can just use the Direct Form 2. – As for band-limited synthesis: it depends on what carrier you actually use and what sample rate you're running on. For relatively low-frequency signals without too much overtone content (e.g. a triangle at 100 Hz), or sample rates above 96 kHz, the difference will hardly be audible. But for, say, a 3 kHz sawtooth with 44.1 kHz sample rate, you will get quite nasty aliasing. You do need a lot of overtones for a vocoder, so you should use band-limited synthesis. $\endgroup$ Commented Jul 27, 2012 at 21:26
  • $\begingroup$ @BigEndian: Have a closer look, the terms are b1 x[n - 1] and -a1 y[n - 1], the first one is applied to the previous input sample, the second one to the previous output sample. Regarding band-limited synthesis, at 44.1kHz, you don't have to go up very high in pitch to notice aliasing on square or sawtooth waves. It just sounds cheap. A good start for an envelope follower would be to take the maximum of the absolute value over windows of 5 or 10 ms, if necessary sent through a 1-pole IIR low-pass. $\endgroup$ Commented Jul 27, 2012 at 21:39
  • $\begingroup$ That would require having a history of tons of inputs, and then every frame you would have to loop through them all just to find the current maximum. Isn't there a faster way? $\endgroup$
    – Void Star
    Commented Jul 27, 2012 at 22:04
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    $\begingroup$ You don't need to store any history to implement the envelope follower I have described. Compare to the local max every sample ; reset the local max and hold its previous value every block. Not sure which branch of DSP you are interested in, but if you want to look into audio stuff, there's a ton of good stuff here: ccrma.stanford.edu/~jos/pubs.html $\endgroup$ Commented Jul 27, 2012 at 22:53

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