Hilbert transform FIR size
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From: Jerry Avins on 8 Aug 2010 07:20 On 8/8/2010 2:56 AM, Erik de Castro Lopo wrote: > jungledmnc wrote: > >> I want to create a frequency shifter for audio. > > Are you sure you want a frequency shifter and not a pitch shifter? > Pitch shifers are much more commonly used for audio than frequency > shifters. > > Pitch shifters do: > > for all Fin : Fout = Fin * a > > which preserves the relationship between harmonics of a single note. > > Frequency shifters do: > > for all Fin : Fout = Fin + a > > which might be useful when used subtly for an audio effect, but will > not give you what you want if what you want is a pitch shifter. In the 1950s, pitch shifting seemed out of the question. Frequency shifting by less than 5 Hz doesn't mess up voice much. Try bringing in a SSB signal using the BFO. There's a wide range that sounds "natural" if you're not familiar with the speaker's true voice. Jerry -- Engineering is the art of making what you want from things you can get. �����������������������������������������������������������������������
From: jungledmnc on 8 Aug 2010 09:02 >On Aug 7, 8:24=A0pm, "jungledmnc" <jungledmnc(a)n_o_s_p_a_m.gmail.com> >wrote: >> I take it there is no simple way to compute the >> allpass coefficients, except for brute-force search. So I did it. > >exactly what? how did you search for your optimal coefs and what did >you get? > >> I wrote a >> simple app, which searches for requested number of allpass sections >> evaluating results like this: >> >> H1 =3D Hallpass_A1(z) * Hallpass_A2(z) * .. >> H2 =3D Hallpass_B1(z) * Hallpass_B2(z) * .. >> where everything is complex. |H1| =3D |H2| =3D 1 as expected. >> >> Finally phase1 =3D atan2(H1.R, H1.I) + PI/2, phase2 =3D atan2(H2.R, H2.I) >> Difference =3D phase1 - phase2, and some wrapping. >> Difference should be as close to 0 as possible. > >difference should be close to 0 or close to pi/2? The PI/2 was added to phase 1 (above). I did it simply by random walk :) - random parameters repeately until it finds some "a little good" solution, then moving the parameters randomly a little searching for better solution. The question is more like, if I multiply the H(z) of each stage and take it's phase, should it represent the resulting phase shift? Or could there be a problem with the measurement by taking an impulse and FFTing the output?
From: jungledmnc on 8 Aug 2010 09:22 Damn it, sorry for my last post, it was a bug, I accidentally added a 1 sample delay to one channel... Such a stupid mistake...
From: Vladimir Vassilevsky on 8 Aug 2010 12:19 jungledmnc wrote: > Damn it, sorry for my last post, it was a bug, I accidentally added a 1 > sample delay to one channel... Such a stupid mistake... Jungledmnc, It is pleasure to see a person who is able to understand the problem and solve this problem, although not in a very elegant way, but quickly, efficiently and entirely by himself. Good luck with your audio work. Vladimir Vassilevsky DSP and Mixed Signal Design Consultant http://www.abvolt.com
From: Angelito Hamm on 8 Aug 2010 12:24
On Aug 6, 5:33 pm, "jungledmnc" <jungledmnc(a)n_o_s_p_a_m.gmail.com> wrote: > Hi, > I want to create a frequency shifter for audio. First I need to get an > analytical signal via a hilbert transformer. I started by checking out how > long the Hilbert FIR would be. Unfortunately I ended with 20ms, which seems > to be related to -3dB at 50Hz (1/0.02). Isn't there another way to do that? > I mean 20ms is a relatively long delay for realtime processing and also 800 > taps would need relatively lots of CPU power. > > Thanks. A detailed description of the pole-zero rotation approach to analytic signal generation using IIR described by Dr. Brackett in this thread http://groups.google.com/group/comp.dsp/msg/08d2545fde320d54?dmode=source is published in Vanbeylen, L. Schoukens, J. "Comparison of Filter Design Methods to generate Analytic Signals" Instrumentation and Measurement Technology Conference, 2006. IMTC 2006. Proceedings of the IEEE 24-27 April 2006 pp. 883 - 887 |