>>
>>Is there a way I can get any gains for the presets? E.g. for a 5-band
>>equalizer, what would be the dB level for each band for a JAZZ preset.
>Or,
>>is it purely aesthetic?
>>
>>Thanks,
>>~ R K
>>
>>
>
>There are no "standards" for centre frequencies or number of bands or Q
>for an Equaliser
>
>Hence I suppose there can be no "standards" for the presets.
>
>So it all just boils down to personal preference.
>
>I have some CDs of 1940-1950's music when Rock and Roll was just getting
>invented. I liked the vocals as in those days, the words still made some
>kind of sense. I would subtly boost the mid to bring up the vocals. That
>was my prefered "Rock and Roll" equaliser setting.
>
>Then I realised that the Bass guitarist was also doing some great stuff.
>But in those days, Bass was not considered a major instrument ( or maybe
>the sound recording and playback systems in those days could not
>capture/playback bass so well as today).
>
>So nowadays, I listen to Early rock and roll with the bass bosted quite
>substantially. My idea of a "Rock and Roll" preset has changed with

time.

>
>Come to think of it, even music mutates and it is hard to exactly
>differentiate early black ethnic from early rock and roll to early rock

to

>fusion rock, jazz rock, punk rock, heavy, acid, death, metal,,,,,,,
>
>So if you cannot exactly define a genre of music, I guess you cannot
>define an exact EQ setting
>
>PS 1: See
>http://www.hydrogenaudio.org/forums/lofiversion/index.php/t30832.html
>
>PS 2: Go to an Audio Shop and copy down the preset settings from one of
>the boxes.
>
>PS 3 : Download Winamp or similar software and see that they do with

their

>EQ pre-sets.
>
>PS 4 : Some users on the internet have made posts requesting Equaliser
>settings saved per song rather than only by Genre. If you are

implementing

>the EQ on a PC, you might consider this idea.
>
>

True. I understand when you say that there are no standards. I was hoping
for any idea. And your suggestions are helpful.
No, I am not implementing the EQ on a PC.
~ Thanks,
R K

Reply by Web Surf●February 23, 20062006-02-23

>
>Is there a way I can get any gains for the presets? E.g. for a 5-band
>equalizer, what would be the dB level for each band for a JAZZ preset.

Or,

>is it purely aesthetic?
>
>Thanks,
>~ R K
>
>

There are no "standards" for centre frequencies or number of bands or Q
for an Equaliser
Hence I suppose there can be no "standards" for the presets.
So it all just boils down to personal preference.
I have some CDs of 1940-1950's music when Rock and Roll was just getting
invented. I liked the vocals as in those days, the words still made some
kind of sense. I would subtly boost the mid to bring up the vocals. That
was my prefered "Rock and Roll" equaliser setting.
Then I realised that the Bass guitarist was also doing some great stuff.
But in those days, Bass was not considered a major instrument ( or maybe
the sound recording and playback systems in those days could not
capture/playback bass so well as today).
So nowadays, I listen to Early rock and roll with the bass bosted quite
substantially. My idea of a "Rock and Roll" preset has changed with time.
Come to think of it, even music mutates and it is hard to exactly
differentiate early black ethnic from early rock and roll to early rock to
fusion rock, jazz rock, punk rock, heavy, acid, death, metal,,,,,,,
So if you cannot exactly define a genre of music, I guess you cannot
define an exact EQ setting
PS 1: See
http://www.hydrogenaudio.org/forums/lofiversion/index.php/t30832.html
PS 2: Go to an Audio Shop and copy down the preset settings from one of
the boxes.
PS 3 : Download Winamp or similar software and see that they do with their
EQ pre-sets.
PS 4 : Some users on the internet have made posts requesting Equaliser
settings saved per song rather than only by Genre. If you are implementing
the EQ on a PC, you might consider this idea.

Reply by rkthebad●February 22, 20062006-02-22

>>>>> Here is one strange thing that I observed. Can anyone explain why?
>>>>>
>>>>> I am trying to see the actual response of the designed 5-band
>>>>> equalizer. I put a gain of +12 dB for all the bands. Then I vary

the

>>>>> sampling frequency of the system. It is observed that at a sampling
>>>>> freq. of 48 kHz, the first band (with fc = 300 Hz) does not show a
>>>>> gain of +12 dB. Instead, it only shows a gain of +9 dB. Why is this
>>>>> so?
>>>>>
>>>>> Thanks,
>>>>> ~ R K
>>
>>
>>I had a similar problem when trying to graph the freq response of R B-J
>>equalisers in Excel.
>>
>>I discovered this problem exists only in High Q Notch/peak filters
>>
>>The reason was that 4 or 5 or 6 or even 10 data points per octave are
>not
>>enough to acurately plot a high Q notch.
>>
>>The answer was to make sure that Fo was one of the calculated and
>plotted
>>data points. It is better if you also calculate and plot some points

+/-

>>2-3% away from Fo
>>
>>( For a multiband Notch Filter, you will need to ensure that each Fo is
>in
>>the list of calculated and plotted frequency points)
>>
>>PS : I can e-mail my Spreadsheet to anyone who is interested.
>>
>>Web Surf
>>
>>
>Could you e-mail me the same please ?
>
>Thanks,
>~ R K
>
>

Thanks a ton to everyone for their previous insights.
Is there a way I can get any gains for the presets? E.g. for a 5-band
equalizer, what would be the dB level for each band for a JAZZ preset. Or,
is it purely aesthetic?
Thanks,
~ R K

Reply by rkthebad●February 21, 20062006-02-21

>>>> Here is one strange thing that I observed. Can anyone explain why?
>>>>
>>>> I am trying to see the actual response of the designed 5-band
>>>> equalizer. I put a gain of +12 dB for all the bands. Then I vary the
>>>> sampling frequency of the system. It is observed that at a sampling
>>>> freq. of 48 kHz, the first band (with fc = 300 Hz) does not show a
>>>> gain of +12 dB. Instead, it only shows a gain of +9 dB. Why is this
>>>> so?
>>>>
>>>> Thanks,
>>>> ~ R K
>
>
>I had a similar problem when trying to graph the freq response of R B-J
>equalisers in Excel.
>
>I discovered this problem exists only in High Q Notch/peak filters
>
>The reason was that 4 or 5 or 6 or even 10 data points per octave are

not

>enough to acurately plot a high Q notch.
>
>The answer was to make sure that Fo was one of the calculated and

plotted

>data points. It is better if you also calculate and plot some points +/-
>2-3% away from Fo
>
>( For a multiband Notch Filter, you will need to ensure that each Fo is

in

>the list of calculated and plotted frequency points)
>
>PS : I can e-mail my Spreadsheet to anyone who is interested.
>
>Web Surf
>
>

Could you e-mail me the same please ?
Thanks,
~ R K

Reply by Web Surf●February 21, 20062006-02-21

>>> Here is one strange thing that I observed. Can anyone explain why?
>>>
>>> I am trying to see the actual response of the designed 5-band
>>> equalizer. I put a gain of +12 dB for all the bands. Then I vary the
>>> sampling frequency of the system. It is observed that at a sampling
>>> freq. of 48 kHz, the first band (with fc = 300 Hz) does not show a
>>> gain of +12 dB. Instead, it only shows a gain of +9 dB. Why is this
>>> so?
>>>
>>> Thanks,
>>> ~ R K

I had a similar problem when trying to graph the freq response of R B-J
equalisers in Excel.
I discovered this problem exists only in High Q Notch/peak filters
The reason was that 4 or 5 or 6 or even 10 data points per octave are not
enough to acurately plot a high Q notch.
The answer was to make sure that Fo was one of the calculated and plotted
data points. It is better if you also calculate and plot some points +/-
2-3% away from Fo
( For a multiband Notch Filter, you will need to ensure that each Fo is in
the list of calculated and plotted frequency points)
PS : I can e-mail my Spreadsheet to anyone who is interested.
Web Surf

Reply by rkthebad●February 21, 20062006-02-21

Is there a way or any article that would suggest as to what should be the
gains for the bands in an equalizer for some presets like ROCK, POP,
etc.?
Thanks,
~ R K

Reply by rkthebad●February 19, 20062006-02-19

>"rkthebad" <raviyenduri@gmail.com> wrote in
>news:vOOdnUD8eYVssmveRVn-sw@giganews.com:
> <********************************************************>
>> Al,
>> You are right. Because of the 16-bit quantization, I am losing
>> precision
>> compared to the floating point architecture. But, the above mentioned
>> problem, I am experiencing even in the floating point implementation.
>> The problem is the same when I tried to compute my coefficients using
>> MATLAB or Microsoft VC++. Is 48 kHz tooo high a sampling freq.
>> compared to the 300 Hz band?
>
>I'm not sure I completely understand your setup.
>
>DFI works best with fixed point assuming you have a double length
>accumulator. In a 16 bit DSP, the accumulator will be 32 bits + guard

bits.

>
>In a 32 bit processor such as a SHARC, you have an 80 bit accumulator

when

>operating in fixed point. This is very good for DFI. I think one of the
>reasons that a lot of DSP books suggest DFII, is that they didn't assume

>the double width accumulator was available. I think this is true of just

>about every DSP, but many of the books were written before there were DSP

>chips.
>
>If you are using IEEE float, you have about 24 bits of mantissa and 8

bits

>of exponent (discounting the hidden bit). The results of the multiplier

and

>accumulator are also IEEE float (24 + 8). There is no extended precision

in

>this mode. DFII might be better for this situation.
>
>My guess is that 300 Hz is too high for 48k sampling with a high Q filter

>with only 16 bits of precision.
>
>Al
>
>
>
>>
>> Thanks,
>> ~ R K
>>
>
>
>
>--
>Al Clark
>Danville Signal Processing, Inc.
>--------------------------------------------------------------------
>Purveyors of Fine DSP Hardware and other Cool Stuff
>Available at http://www.danvillesignal.com
>

That might explain why the DF1 structure on floating point is not giving
me good results. I need some time to finish working on this. I shall get
back to you guys for any further help.
Thanks,
~ R K

Reply by Al Clark●February 17, 20062006-02-17

"rkthebad" <raviyenduri@gmail.com> wrote in
news:vOOdnUD8eYVssmveRVn-sw@giganews.com:

>>"rkthebad" <raviyenduri@gmail.com> wrote in
>>news:J8KdncyPSKDCkGveRVn-ug@giganews.com:
>>
>>>>
>>>>rkthebad wrote:
>>>>
>>>>>
>>>>> Apart from the conditions that you mentioned above, I found one
>>>>> more stability condition in a textbook.
>>>>> Condition is : |a2| < 1 and |a1| < 1 + a2.
>>>>> I did not venture too much into the math, but I am guessing your
>>>>> conditions might come down to this after simplification.
>>>>
>>>>the second one looks different, but the first is the same.
>>>>
>>>>whatever, i am convinced that the criteria i stated:
>>>>
>>>> 1. if (a1/2)^2 >= a2 then |a1/2| + sqrt((a1/2)^2 - a2) < 1
>>>>
>>>> 2. if (a1/2)^2 < a2 then a2 < 1
>>>>
>>>>is both necessary and sufficient.
>>>>
>>>>if it's case 1. then
>>>>
>>>> sqrt((a1/2)^2 - a2) < 1 - |a1/2|
>>>>
>>>> (a1/2)^2 - a2 < (1 - |a1/2|)^2 = 1 - |a1| + (a1/2)^2
>>>>
>>>>which comes to
>>>>
>>>> |a1| < 1 + a2
>>>>
>>>>so you're right, it's the same thing (except, it's a "either or" not
>>>>*both*). thanks for pointing that out.
>>>>
>>>>> But, wouldn't it be sufficient to see whether the poles of the
>>> quantized
>>>>> filter are inside the unit circle?
>>>>
>>>>that's precisely what i was doing. take a look at the math, that is
>>>>all that it is.
>>>>
>>>>r b-j
>>>>
>>>>
>>> Here is one strange thing that I observed. Can anyone explain why?
>>>
>>> I am trying to see the actual response of the designed 5-band
>>> equalizer. I put a gain of +12 dB for all the bands. Then I vary the
>>> sampling frequency of the system. It is observed that at a sampling
>>> freq. of 48 kHz, the first band (with fc = 300 Hz) does not show a
>>> gain of +12 dB. Instead, it only shows a gain of +9 dB. Why is this
>>> so?
>>>
>>> Thanks,
>>> ~ R K
>>>
>>>
>>
>>I think you are a victim of 16 bit precision. The low frequency bands
>>with high Q are the first place where the problem is likely to occur.
>>
>>This kind of problem illustrates why the Motorola 56K (24 bits) and
>>later
>
>>the SHARC (32 bits) became so popular in high performance audio.
>>
>>I think you already found improvements by changing filter topology
>>(DFI vs DF2). If you have enough MIPs, you can rewrite the algorithm
>>with double precision math with your 16 bit processor or move to a
>>higher precision processor.
>>
>>
>>
>>
>>
>>--
>>Al Clark
>>Danville Signal Processing, Inc.
>>--------------------------------------------------------------------
>>Purveyors of Fine DSP Hardware and other Cool Stuff
>>Available at http://www.danvillesignal.com
>>
> Al,
> You are right. Because of the 16-bit quantization, I am losing
> precision
> compared to the floating point architecture. But, the above mentioned
> problem, I am experiencing even in the floating point implementation.
> The problem is the same when I tried to compute my coefficients using
> MATLAB or Microsoft VC++. Is 48 kHz tooo high a sampling freq.
> compared to the 300 Hz band?

I'm not sure I completely understand your setup.
DFI works best with fixed point assuming you have a double length
accumulator. In a 16 bit DSP, the accumulator will be 32 bits + guard bits.
In a 32 bit processor such as a SHARC, you have an 80 bit accumulator when
operating in fixed point. This is very good for DFI. I think one of the
reasons that a lot of DSP books suggest DFII, is that they didn't assume
the double width accumulator was available. I think this is true of just
about every DSP, but many of the books were written before there were DSP
chips.
If you are using IEEE float, you have about 24 bits of mantissa and 8 bits
of exponent (discounting the hidden bit). The results of the multiplier and
accumulator are also IEEE float (24 + 8). There is no extended precision in
this mode. DFII might be better for this situation.
My guess is that 300 Hz is too high for 48k sampling with a high Q filter
with only 16 bits of precision.
Al

>
> Thanks,
> ~ R K
>

--
Al Clark
Danville Signal Processing, Inc.
--------------------------------------------------------------------
Purveyors of Fine DSP Hardware and other Cool Stuff
Available at http://www.danvillesignal.com

Reply by rkthebad●February 17, 20062006-02-17

>>"rkthebad" <raviyenduri@gmail.com> wrote in
>>news:J8KdncyPSKDCkGveRVn-ug@giganews.com:
>>
>>>>
>>>>rkthebad wrote:
>>>>
>>>>>
>>>>> Apart from the conditions that you mentioned above, I found one

more

>>>>> stability condition in a textbook.
>>>>> Condition is : |a2| < 1 and |a1| < 1 + a2.
>>>>> I did not venture too much into the math, but I am guessing your
>>>>> conditions might come down to this after simplification.
>>>>
>>>>the second one looks different, but the first is the same.
>>>>
>>>>whatever, i am convinced that the criteria i stated:
>>>>
>>>> 1. if (a1/2)^2 >= a2 then |a1/2| + sqrt((a1/2)^2 - a2) < 1
>>>>
>>>> 2. if (a1/2)^2 < a2 then a2 < 1
>>>>
>>>>is both necessary and sufficient.
>>>>
>>>>if it's case 1. then
>>>>
>>>> sqrt((a1/2)^2 - a2) < 1 - |a1/2|
>>>>
>>>> (a1/2)^2 - a2 < (1 - |a1/2|)^2 = 1 - |a1| + (a1/2)^2
>>>>
>>>>which comes to
>>>>
>>>> |a1| < 1 + a2
>>>>
>>>>so you're right, it's the same thing (except, it's a "either or" not
>>>>*both*). thanks for pointing that out.
>>>>
>>>>> But, wouldn't it be sufficient to see whether the poles of the
>>> quantized
>>>>> filter are inside the unit circle?
>>>>
>>>>that's precisely what i was doing. take a look at the math, that is
>>>>all that it is.
>>>>
>>>>r b-j
>>>>
>>>>
>>> Here is one strange thing that I observed. Can anyone explain why?
>>>
>>> I am trying to see the actual response of the designed 5-band
>>> equalizer. I put a gain of +12 dB for all the bands. Then I vary the
>>> sampling frequency of the system. It is observed that at a sampling
>>> freq. of 48 kHz, the first band (with fc = 300 Hz) does not show a
>>> gain of +12 dB. Instead, it only shows a gain of +9 dB. Why is this
>>> so?
>>>
>>> Thanks,
>>> ~ R K
>>>
>>>
>>
>>I think you are a victim of 16 bit precision. The low frequency bands
>>with high Q are the first place where the problem is likely to occur.
>>
>>This kind of problem illustrates why the Motorola 56K (24 bits) and

later

>
>>the SHARC (32 bits) became so popular in high performance audio.
>>
>>I think you already found improvements by changing filter topology (DFI

>>vs DF2). If you have enough MIPs, you can rewrite the algorithm with
>>double precision math with your 16 bit processor or move to a higher
>>precision processor.
>>
>>
>>
>>
>>
>>--
>>Al Clark
>>Danville Signal Processing, Inc.
>>--------------------------------------------------------------------
>>Purveyors of Fine DSP Hardware and other Cool Stuff
>>Available at http://www.danvillesignal.com
>>
>Al,
> You are right. Because of the 16-bit quantization, I am losing

precision

>compared to the floating point architecture. But, the above mentioned
>problem, I am experiencing even in the floating point implementation.

The

>problem is the same when I tried to compute my coefficients using MATLAB
>or Microsoft VC++. Is 48 kHz tooo high a sampling freq. compared to the
>300 Hz band?
>
>Thanks,
>~ R K
>
>

Everyone,
I am able to get a *decent* output through my 5-band equalizer. But, I
am sacrificing the bass content till 500 Hz. Now, I know for sure that
there is serious problem with a high sampling frequency and my first band
(if below 500 Hz). I would have to find a way around this. Any suggestions
will be definitely helpful.
Thanks to everyone.
~ R K

Reply by rkthebad●February 17, 20062006-02-17

>"rkthebad" <raviyenduri@gmail.com> wrote in
>news:J8KdncyPSKDCkGveRVn-ug@giganews.com:
>
>>>
>>>rkthebad wrote:
>>>
>>>>
>>>> Apart from the conditions that you mentioned above, I found one more
>>>> stability condition in a textbook.
>>>> Condition is : |a2| < 1 and |a1| < 1 + a2.
>>>> I did not venture too much into the math, but I am guessing your
>>>> conditions might come down to this after simplification.
>>>
>>>the second one looks different, but the first is the same.
>>>
>>>whatever, i am convinced that the criteria i stated:
>>>
>>> 1. if (a1/2)^2 >= a2 then |a1/2| + sqrt((a1/2)^2 - a2) < 1
>>>
>>> 2. if (a1/2)^2 < a2 then a2 < 1
>>>
>>>is both necessary and sufficient.
>>>
>>>if it's case 1. then
>>>
>>> sqrt((a1/2)^2 - a2) < 1 - |a1/2|
>>>
>>> (a1/2)^2 - a2 < (1 - |a1/2|)^2 = 1 - |a1| + (a1/2)^2
>>>
>>>which comes to
>>>
>>> |a1| < 1 + a2
>>>
>>>so you're right, it's the same thing (except, it's a "either or" not
>>>*both*). thanks for pointing that out.
>>>
>>>> But, wouldn't it be sufficient to see whether the poles of the
>> quantized
>>>> filter are inside the unit circle?
>>>
>>>that's precisely what i was doing. take a look at the math, that is
>>>all that it is.
>>>
>>>r b-j
>>>
>>>
>> Here is one strange thing that I observed. Can anyone explain why?
>>
>> I am trying to see the actual response of the designed 5-band
>> equalizer. I put a gain of +12 dB for all the bands. Then I vary the
>> sampling frequency of the system. It is observed that at a sampling
>> freq. of 48 kHz, the first band (with fc = 300 Hz) does not show a
>> gain of +12 dB. Instead, it only shows a gain of +9 dB. Why is this
>> so?
>>
>> Thanks,
>> ~ R K
>>
>>
>
>I think you are a victim of 16 bit precision. The low frequency bands
>with high Q are the first place where the problem is likely to occur.
>
>This kind of problem illustrates why the Motorola 56K (24 bits) and later

>the SHARC (32 bits) became so popular in high performance audio.
>
>I think you already found improvements by changing filter topology (DFI
>vs DF2). If you have enough MIPs, you can rewrite the algorithm with
>double precision math with your 16 bit processor or move to a higher
>precision processor.
>
>
>
>
>
>--
>Al Clark
>Danville Signal Processing, Inc.
>--------------------------------------------------------------------
>Purveyors of Fine DSP Hardware and other Cool Stuff
>Available at http://www.danvillesignal.com
>

Al,
You are right. Because of the 16-bit quantization, I am losing precision
compared to the floating point architecture. But, the above mentioned
problem, I am experiencing even in the floating point implementation. The
problem is the same when I tried to compute my coefficients using MATLAB
or Microsoft VC++. Is 48 kHz tooo high a sampling freq. compared to the
300 Hz band?
Thanks,
~ R K