Does anyone have a set of IIR coefficients that mimick the A-weighting filter used for audio SNR measurements? I'm sure it's been done many times but I'm not able to find it on the WEB. Thanks! Bob Adams
Coefficients for A-weighting filter
Started by ●December 15, 2006
Reply by ●December 15, 20062006-12-15
"Robert Adams" <robert.adams@analog.com> wrote in news:1166220102.470664.17450@80g2000cwy.googlegroups.com:> Does anyone have a set of IIR coefficients that mimick the A-weighting > filter used for audio SNR measurements? I'm sure it's been done many > times but I'm not able to find it on the WEB. > > > Thanks! > > > > Bob Adams > >Bob, I don't think you will find it on the web. I have done this search more than once. The problem is the 12200 Hz corners since the sampling rate is never quite high enough for the bilinear transform to work without warping getting in the way. What sample rate are you using? I may have some coefficients that were derived using a curve fitting program. I have started a process to solve this problem about 5 times but I never get around to finishing it. If you solve it, send me the results, please..... -- Al Clark Danville Signal Processing, Inc. -------------------------------------------------------------------- Purveyors of Fine DSP Hardware and other Cool Stuff Available at http://www.danvillesignal.com
Reply by ●December 15, 20062006-12-15
On 15 Dec 2006 14:01:42 -0800, "Robert Adams" <robert.adams@analog.com> wrote:>Does anyone have a set of IIR coefficients that mimick the A-weighting >filter used for audio SNR measurements?You can make your own. If you (or anyone else) wish, I can send you a copy of my Matlab frequency-domain least-squares filter design program. All you have to supply is sampling rate and a bunch of target magnitude/phase frequency response measurements. See http://groups.google.com/group/comp.dsp/browse_frm/thread/501b31898ae74f47 particularly message 21 and onward. Email if interested; remove the "i" from "comicast" in my address. Greg Berchin
Reply by ●December 15, 20062006-12-15
On Fri, 15 Dec 2006 22:13:35 GMT, Al Clark <dsp@danvillesignal.com> wrote:>If you solve it, send me the results, please.....Al, the latest version of the FDLS code (available from me) works better than the version I sent to you so many years ago (bug fixes). And it's written in Matlab. Email if interested. Remove the "i" from "comicast". Greg
Reply by ●December 15, 20062006-12-15
Robert Adams wrote:> Does anyone have a set of IIR coefficients that mimick the A-weighting > filter used for audio SNR measurements? I'm sure it's been done many > times but I'm not able to find it on the WEB.hey Bob, this ain't the A-weighting another approximation that's supposed to be better. the Wannamaker/Lipshitz "F-wieghting" curve from their "Psychoacoutically Optimal Noise-Shaping" paper in some old AES journal. it is defined below (as an s-plane filter representing the inverse of the 0 dB Fletcher-Munson curve, if you want the equal loudness curve, swap the poles and zeros, reciprocate the constant gain g, and plot the magnitude): 12 zeros, 55 poles: z1 = 0.0 four z1 zeros z2 = -0.58 +/- j*1.03 one z2 pair of zeros z3 = -3.18 +/- j*8.75 three z3 pairs of zeros p1 = -0.18 three p1 poles p2 = -1.63 two p2 poles p3 = -2.51 +/- j*3.85 four p3 pairs of poles p4 = -6.62 +/- j*14.29 twenty p4 pairs of poles g = 6.727242106827342e+47; g = constant gain to normalize 1 kHz to 0 dB all poles and zeros are represented in kHz (not krad/sec so they really have to multiplied be 2*pi and perhaps 1000 to be normal s-plane poles and zeros with the second as the unit time). r b-j
Reply by ●December 15, 20062006-12-15
Greg Berchin <gberchin@comicast.net> wrote in news:ed76o21tcrrcqmfhmf797m3qqc9062qjtk@4ax.com:> On 15 Dec 2006 14:01:42 -0800, "Robert Adams" > <robert.adams@analog.com> wrote: > >>Does anyone have a set of IIR coefficients that mimick the A-weighting >>filter used for audio SNR measurements? > > You can make your own. If you (or anyone else) wish, I can send you a > copy of my Matlab frequency-domain least-squares filter design > program. All you have to supply is sampling rate and a bunch of target > magnitude/phase frequency response measurements. See > http://groups.google.com/group/comp.dsp/browse_frm/thread/501b31898ae74 > f47 particularly message 21 and onward. > > Email if interested; remove the "i" from "comicast" in my address. > > Greg Berchin >The fit I was talking about was Greg's work -- Al Clark Danville Signal Processing, Inc. -------------------------------------------------------------------- Purveyors of Fine DSP Hardware and other Cool Stuff Available at http://www.danvillesignal.com
Reply by ●December 15, 20062006-12-15
"robert bristow-johnson" <rbj@audioimagination.com> wrote in news:1166222409.129012.96080@16g2000cwy.googlegroups.com:> Robert Adams wrote: >> Does anyone have a set of IIR coefficients that mimick the A-weighting >> filter used for audio SNR measurements? I'm sure it's been done many >> times but I'm not able to find it on the WEB. > > hey Bob, this ain't the A-weighting another approximation that's > supposed to be better. the Wannamaker/Lipshitz "F-wieghting" curve > from their "Psychoacoutically Optimal Noise-Shaping" paper in some old > AES journal. > > it is defined below (as an s-plane filter representing the inverse of > the 0 dB Fletcher-Munson curve, if you want the equal loudness curve, > swap the poles and zeros, reciprocate the constant gain g, and plot the > magnitude): > > 12 zeros, 55 poles: > > z1 = 0.0 four z1 zeros > z2 = -0.58 +/- j*1.03 one z2 pair of zeros > z3 = -3.18 +/- j*8.75 three z3 pairs of zeros > > p1 = -0.18 three p1 poles > p2 = -1.63 two p2 poles > p3 = -2.51 +/- j*3.85 four p3 pairs of poles > p4 = -6.62 +/- j*14.29 twenty p4 pairs of poles > > g = 6.727242106827342e+47; > g = constant gain to normalize 1 kHz to 0 dB > > all poles and zeros are represented in kHz (not krad/sec so they really > have to multiplied be 2*pi and perhaps 1000 to be normal s-plane poles > and zeros with the second as the unit time). > > r b-j > >The A weight curve is not wonderful but it is the world's defacto measurement standard. It is defined exactly (at least in acoustics) as an s-plane filter. This is why it is can't be expressed perfectly as a digital filter. -- Al Clark Danville Signal Processing, Inc. -------------------------------------------------------------------- Purveyors of Fine DSP Hardware and other Cool Stuff Available at http://www.danvillesignal.com
Reply by ●December 15, 20062006-12-15
Yes, the reason I'm interested is that I want to predict what the datasheet spec will be on a product I'm designing, so even though A-weighting is outdated, it does the job for me. I'll work on this and post the results; I was just hoping I could be lazy! Does anyone have the S-plane poles/zeros? Regards Bob Al Clark wrote:> "robert bristow-johnson" <rbj@audioimagination.com> wrote in > news:1166222409.129012.96080@16g2000cwy.googlegroups.com: > > > Robert Adams wrote: > >> Does anyone have a set of IIR coefficients that mimick the A-weighting > >> filter used for audio SNR measurements? I'm sure it's been done many > >> times but I'm not able to find it on the WEB. > > > > hey Bob, this ain't the A-weighting another approximation that's > > supposed to be better. the Wannamaker/Lipshitz "F-wieghting" curve > > from their "Psychoacoutically Optimal Noise-Shaping" paper in some old > > AES journal. > > > > it is defined below (as an s-plane filter representing the inverse of > > the 0 dB Fletcher-Munson curve, if you want the equal loudness curve, > > swap the poles and zeros, reciprocate the constant gain g, and plot the > > magnitude): > > > > 12 zeros, 55 poles: > > > > z1 = 0.0 four z1 zeros > > z2 = -0.58 +/- j*1.03 one z2 pair of zeros > > z3 = -3.18 +/- j*8.75 three z3 pairs of zeros > > > > p1 = -0.18 three p1 poles > > p2 = -1.63 two p2 poles > > p3 = -2.51 +/- j*3.85 four p3 pairs of poles > > p4 = -6.62 +/- j*14.29 twenty p4 pairs of poles > > > > g = 6.727242106827342e+47; > > g = constant gain to normalize 1 kHz to 0 dB > > > > all poles and zeros are represented in kHz (not krad/sec so they really > > have to multiplied be 2*pi and perhaps 1000 to be normal s-plane poles > > and zeros with the second as the unit time). > > > > r b-j > > > > > > The A weight curve is not wonderful but it is the world's defacto > measurement standard. It is defined exactly (at least in acoustics) as an > s-plane filter. This is why it is can't be expressed perfectly as a > digital filter. > > > -- > Al Clark > Danville Signal Processing, Inc. > -------------------------------------------------------------------- > Purveyors of Fine DSP Hardware and other Cool Stuff > Available at http://www.danvillesignal.com
Reply by ●December 15, 20062006-12-15
On 15 Dec 2006 16:46:49 -0800, "Robert Adams" <robert.adams@analog.com> wrote:>Does anyone have the S-plane poles/zeros?http://www.ptpart.co.uk/show.php?contentid=70
Reply by ●December 16, 20062006-12-16
Greg Berchin wrote:> On 15 Dec 2006 16:46:49 -0800, "Robert Adams" <robert.adams@analog.com> > wrote: > > >Does anyone have the S-plane poles/zeros? > > http://www.ptpart.co.uk/show.php?contentid=70that's useful, Greg. it oughta be okay to just apply the BLT* to the poles/zeros to get a digital filter. r b-j *(bacon, lettuce, and tomato sandwich, i prefer it on rye)
