What would be the most effecient method of hilbert transofrming audio? my filter program genertates too many taps for the low frequency performacnce i require. I don't want to use FFT due to memory constraints in the DSP. Thanks in advance

# most effecient hilbert transform method

Started by ●June 4, 2004

Reply by ●June 4, 20042004-06-04

scott@scottincz.com wrote:> What would be the most effecient method of hilbert transofrming audio? > my filter program genertates too many taps for the low frequency > performacnce i require. I don't want to use FFT due to memory > constraints in the DSP. > Thanks in advanceDo you really need to generate a signal with every frequency component 90 degrees shifted from a delayed replica of the original, ot id it sufficient to produce a pair of signals, one of which is in quadrature with the other, neither bearing a simple phase relationship to the input? The second version is easier to achieve. It uses a pair of bandpass filters generated from a low-pass prototype by multiplying its coefficients in one case with a sine, and the other with the cosine od the bandpass's center frequency. It is convenient to make that frequency Fs/4. Jerry -- Engineering is the art of making what you want from things you can get. �����������������������������������������������������������������������

Reply by ●June 4, 20042004-06-04

On Fri, 04 Jun 2004 17:56:39 -0400, Jerry Avins <jya@ieee.org> wrote:>scott@scottincz.com wrote: > >> What would be the most effecient method of hilbert transofrming audio? >> my filter program genertates too many taps for the low frequency >> performacnce i require. I don't want to use FFT due to memory >> constraints in the DSP. >> Thanks in advance > >Do you really need to generate a signal with every frequency component >90 degrees shifted from a delayed replica of the original, ot id it >sufficient to produce a pair of signals, one of which is in quadrature >with the other, neither bearing a simple phase relationship to the >input? The second version is easier to achieve. It uses a pair of >bandpass filters generated from a low-pass prototype by multiplying its >coefficients in one case with a sine, and the other with the cosine od >the bandpass's center frequency. It is convenient to make that frequency >Fs/4. > >Jerryideally 20hz-20khz so the whole audio range.

Reply by ●June 4, 20042004-06-04

scott@scottincz.com wrote:> On Fri, 04 Jun 2004 17:56:39 -0400, Jerry Avins <jya@ieee.org> wrote: > > >>scott@scottincz.com wrote: >> >> >>>What would be the most effecient method of hilbert transofrming audio? >>>my filter program genertates too many taps for the low frequency >>>performacnce i require. I don't want to use FFT due to memory >>>constraints in the DSP. >>>Thanks in advance >> >>Do you really need to generate a signal with every frequency component >>90 degrees shifted from a delayed replica of the original, ot id it >>sufficient to produce a pair of signals, one of which is in quadrature >>with the other, neither bearing a simple phase relationship to the >>input? The second version is easier to achieve. It uses a pair of >>bandpass filters generated from a low-pass prototype by multiplying its >>coefficients in one case with a sine, and the other with the cosine od >>the bandpass's center frequency. It is convenient to make that frequency >>Fs/4. >> >>Jerry > > > > ideally 20hz-20khz so the whole audio range.That's not what I asked. your original sampled signal is good from DC to Fs/2. You can pass that signal through two bandpass filters (they share a data buffer) centered about Fs/4 with passbands from 20 Hz to Fs-20 Hz with less computation than needed by a single Hilbert transformer. Unlike the HT approach, this one doesn't preserve the original phase. Jerry -- Engineering is the art of making what you want from things you can get. �����������������������������������������������������������������������

Reply by ●June 5, 20042004-06-05

Jerry Avins wrote:> scott@scottincz.com wrote: > >> What would be the most effecient method of hilbert transofrming audio? >> my filter program genertates too many taps for the low frequency >> performacnce i require. I don't want to use FFT due to memory >> constraints in the DSP. >> Thanks in advance > > > Do you really need to generate a signal with every frequency component > 90 degrees shifted from a delayed replica of the original, ot id it > sufficient to produce a pair of signals, one of which is in quadrature > with the other, neither bearing a simple phase relationship to the > input? The second version is easier to achieve. It uses a pair of > bandpass filters generated from a low-pass prototype by multiplying its > coefficients in one case with a sine, and the other with the cosine od > the bandpass's center frequency. It is convenient to make that frequency > Fs/4. > > JerryOr if you _really_ don't mind mucking up the phase you could use IIR all-pass filters. It'd take some dinking with the math and you'd have to be careful about realizing them to avoid numeric problems, but you could do it with a relative minimum of processing power. -- Tim Wescott Wescott Design Services http://www.wescottdesign.com

Reply by ●June 5, 20042004-06-05

Reply by ●June 5, 20042004-06-05

David Joseph Bonnici wrote:> I suggest the use of the following. > http://www.nauticom.net/www/jdtaft/special_fir.htm > > DavidOP wants not a HT, but something to do his job faster. We're trying to find out how much of what else he can give up to get it. Thanks for the link. It's a nice one. Jerry -- Engineering is the art of making what you want from things you can get. �����������������������������������������������������������������������

Reply by ●June 5, 20042004-06-05

On Sat, 05 Jun 2004 11:07:33 -0400, Jerry Avins <jya@ieee.org> wrote:>David Joseph Bonnici wrote: > >> I suggest the use of the following. >> http://www.nauticom.net/www/jdtaft/special_fir.htm >> >> David > >OP wants not a HT, but something to do his job faster. We're trying to >find out how much of what else he can give up to get it.What i require is good amplitude performance at low frequencies, say 30-50 Hz but i also need the audio to extend up to 20khz. At the moment the taps come out in the thousands for the performance i am after. The hardware won't allow FFT. The application is the removal of the lower sideband in an AM modulator.> >Thanks for the link. It's a nice one. > >Jerry

Reply by ●June 5, 20042004-06-05

scott@scottincz.com wrote:> On Sat, 05 Jun 2004 11:07:33 -0400, Jerry Avins <jya@ieee.org> wrote: > > >>David Joseph Bonnici wrote: >> >> >>>I suggest the use of the following. >>>http://www.nauticom.net/www/jdtaft/special_fir.htm >>> >>>David >> >>OP wants not a HT, but something to do his job faster. We're trying to >>find out how much of what else he can give up to get it. > > > What i require is good amplitude performance at low frequencies, say > 30-50 Hz but i also need the audio to extend up to 20khz. At the > moment the taps come out in the thousands for the performance i am > after. The hardware won't allow FFT. > The application is the removal of the lower sideband in an AM > modulator. > > > >>Thanks for the link. It's a nice one. >> >>JerryIf you want real numbers, you'll have to divulge the sample rate and the passband of the anti-alias filter. Jerry -- Engineering is the art of making what you want from things you can get. �����������������������������������������������������������������������

Reply by ●June 5, 20042004-06-05