Hi, I would like to generate an audio clip with a varying frequency. A few attempts using either matlab/octave or Python code have made "interesting" sounds but not what I'm looking for yet. The sound I want will have an initial frequency F0 which will shift (interpolate) to a frequency F1 midway through the clip, and remain at F1 until the end. For example, in a two-second clip the frequency starts at 110Hz and smoothly increases to 220Hz at the 1-second point. 220Hz is maintained to the clip end at 2 seconds. I've tried just changing the frequency in sin(2*pi*F*t) w.r.t. the time parameter t, but the resulting wave is discontinuous, buzzy, etc. I've also tried using FM sythesis but I don't understand enough about how the I and Fm terms behave with the phase shifting (as in sin(2*pi*Fc*t + I*sin(2*pi*Fm*t))). What am I doing wrong, or what techniques would make this easier? If FM sythesis can be used, what values for Fc,I,Fm are suggested for the clip described above? If you have sample code that shows how to do this, I would appreciate seeing it (c,c++,python,matlab, etc). thanks, Brent p.s. Sample Python code: # import sys, math, wave, array def sourceFM(Fc, t, I, Fm): return math.sin(twopi * Fc * t + I * math.sin(twopi * Fm * t)) twopi = 2.0 * math.pi Fs = 44100 secs = 2.0 numsamples = int(Fs * secs) samples = [0.0] * numsamples Fc = 110 I = 110 Fm = 0.25 print "Generating...%d samples" % (numsamples) for i in range(numsamples): samples[i] = sourceFM(Fc, float(i)/numsamples, I, Fm) # ...rest of code normalized, converted to 16-bit ints, # and wrote a .wav file. # - EOF

# Audio synthesis problem

Started by ●January 4, 2007

Reply by ●January 4, 20072007-01-04

Brent wrote:> Hi, > > I would like to generate an audio clip with a varying > frequency. A few attempts using either matlab/octave or > Python code have made "interesting" sounds but not what > I'm looking for yet. > > The sound I want will have an initial frequency F0 which > will shift (interpolate) to a frequency F1 midway through > the clip, and remain at F1 until the end. > For example, in a two-second clip the frequency starts > at 110Hz and smoothly increases to 220Hz at the 1-second > point. 220Hz is maintained to the clip end at 2 seconds. > > I've tried just changing the frequency in sin(2*pi*F*t) > w.r.t. the time parameter t, but the resulting wave > is discontinuous, buzzy, etc. > > I've also tried using FM sythesis but I don't understand > enough about how the I and Fm terms behave with the phase > shifting (as in sin(2*pi*Fc*t + I*sin(2*pi*Fm*t))). > > What am I doing wrong, or what techniques would make this > easier? > If FM sythesis can be used, what values for Fc,I,Fm are > suggested for the clip described above? > If you have sample code that shows how to do this, I > would appreciate seeing it (c,c++,python,matlab, etc). > > thanks, > Brent > > p.s. Sample Python code: > # > import sys, math, wave, array > > def sourceFM(Fc, t, I, Fm): > return math.sin(twopi * Fc * t + > I * math.sin(twopi * Fm * t)) > > twopi = 2.0 * math.pi > Fs = 44100 > secs = 2.0 > numsamples = int(Fs * secs) > > samples = [0.0] * numsamples > Fc = 110 > I = 110 > Fm = 0.25 > print "Generating...%d samples" % (numsamples) > for i in range(numsamples): > samples[i] = sourceFM(Fc, float(i)/numsamples, I, Fm) > > # ...rest of code normalized, converted to 16-bit ints, > # and wrote a .wav file. > # - EOFYou need to integrate your phase step-by-step. I.e., for each sample: * figure out the frequency you want * set phase = phase + frequency * deltaT * calculate x = sin(phase) where phase is the integral of frequency, and deltaT is your sampling interval (i.e. 1/44000 if you're sampling at 44000Hz). -- Tim Wescott Wescott Design Services http://www.wescottdesign.com Posting from Google? See http://cfaj.freeshell.org/google/ "Applied Control Theory for Embedded Systems" came out in April. See details at http://www.wescottdesign.com/actfes/actfes.html

Reply by ●January 4, 20072007-01-04

Tim Wescott wrote:> Brent wrote: > >> Hi, >> >> I would like to generate an audio clip ... >> ... > > You need to integrate your phase step-by-step. I.e., for each sample: > > * figure out the frequency you want > * set phase = phase + frequency * deltaT > * calculate x = sin(phase) > > where phase is the integral of frequency, and deltaT is your sampling > interval (i.e. 1/44000 if you're sampling at 44000Hz). >Thank you. I tried to track phase in one of my earlier non-FM attempts and danced all around it, but had it wrong. -Brent