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  Q1.15 calculation         


Author: Frank
Date: Jun 8, 2010 11:49

Hey guys,

I was wondering if you could help me optimize the code below (if possible):

R = (B*X);
tmp = (int16_t) (R>>15);
R = ((int32_t)tmp)+Z;
tmp = (int16_t)R;
R = (A*((int32_t)Y));
tmp2 = -(int16_t) (R>>15);
Rout=tmp+tmp2;

Rout is a SINT16 variable containing a Q1.15 value
A and B will be replaced by a number between -32768 and 32767
tmp and tmp2 are SINT16 variables; each containing a Q1.15 value
Z is a SINT32 variable containing a Q1.15 value
Y is a SINT16 variable containing a Q1.15 value
R is a SINT32 variable containing the result of an operation on two Q1.15
values
X is a SINT32 variable containing a Q1.15 value
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  Re: kalman filter to improve the gps data received from an iPhone         


Author: claudegps
Date: Jun 8, 2010 10:19

On 25 Mag, 16:28, "andreivig" n_o_s_p_a_m.gmail.com>
wrote:
> Hi,
>
> I am currently working on an navigation application for an iPhone device.
> The GPS data that I receive is not very accurate and I want to use a kalman
> filter. The problem is that I don't really know how to build the kalman
> equations for x and y coordinates. What kind of model should I use?

Now iPhone4 has also the gyro!
3 Axis accelerometer + 3 axis gyro and you have your IMU :)
no comments
  Demodulator - Slicing by float to fixed conversion         


Author: m26k9
Date: Jun 8, 2010 07:50

Hello,

I had a thread earlier about floating-point to fixed-point conversion for a
confusion I had with the QAM demodulator code I have (SHARC dsp). This is
my first time working with an demodulator assembly and it has really thrown
me off. I am trying to figure out how the slicing is performed. If anybody
could give any hint that would be great.

1) What I am confused is, how floating-point to fixed-point conversion can
perform the slicing, because, for example, in QAM, the point 3, can be
received 2.4. Conversion to fixed-point puts it at 2, which is not a valid
QAM point?

2) And it is mentioned that this floating-point to fixed-point conversion
will put the MSB of the constellation at bit 31 to due proper scaling.

The constellation gains are applied to the fft output before the slicing.
So my guess is that the floating point value will be something +/- 1.xxx,
3.xxx, 5xxx, etc. Where the .xxx are fractional points coming from noise.
The noise will very well take them over the boundaries to 2.xxx, 4.xxx
too.
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  See Hot Sexy Star Priyamani Nude Bathing Videos In All Angles.         


Author: KAJOL
Date: Jun 8, 2010 07:07

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  Kenlighten - A social network for knowledge seekers and providers         


Author: uma
Date: Jun 8, 2010 07:06

Dear friends,

I am writing to share with you about a unique social knowledge network
- Kenlighten http://www.kenlighten.com .

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tutorials in your area of expertise by uploading your ebooks,
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your connections.

So do visit http://www.kenlighten.com and subscribe for your free
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acquaintances to connect with you at Kenlighten.

Happy networking and knowledge sharing!

Uma
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  Simple hack to get $800 to your home.         


Author: money mania
Date: Jun 8, 2010 06:26

Simple hack to get $800 to your home at http://mastidunia.tk

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cheque.please dont tell to anyone.
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  audio DSP SIG in Seattle         


Author: Pion
Date: Jun 8, 2010 06:22

I am wondering if there is any audio DSP special interest group (SIG)
in Seattle.

Any pointer is appreciated.

Thanks in advance for your help.
no comments
  inverse z-transform - partial fractions         


Author: bos1234
Date: Jun 8, 2010 06:09

2 Comments
  Re: Recalculating difference equation coefficients when the time base changes         


Author: Tim Wescott
Date: Jun 8, 2010 05:36

On 06/06/2010 01:02 PM, pnachtwey wrote:
> On Jun 4, 2:28 pm, Tim Wescott wrote:
>> On 06/03/2010 11:21 PM, pnachtwey wrote:
>>
>>
>>
>>
>>
>>> On Jun 3, 5:42 pm, Fred Marshallremove_the_xacm.org>
>>> wrote:
>>>> pnachtwey wrote:
>>>>> I have a difference equation
>>>>> y(n)=A1*y(n-1)+A2*y(n-2)...+B1*x(n-1)+B2*Y(n-2)…+C
>>>>> The coefficients A1,A2,...B1,B2,.... were computed with a known sample
>>>>> time. C is a null offset.
>>>>> Is there an easy way to recalculate A1,A2,..,B1,B2,... for another
>>>>> sample time so the response is the same?
>>>>> Doing this for something simple like
>>>>> y(n)=A1*y(n-1)+B1*x(n-1)+C
>>>>> A1new=A1old^(Tnew/Told) ...
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  Re: Converting IIR filter to different sample rates         


Author: pnachtwey
Date: Jun 8, 2010 03:57

On Jun 7, 5:18 am, "PaulTapper" n_o_s_p_a_m.hotmail.com>
wrote:
> Hi,
>
> Is there a standard way of converting an IIR filter to a different sample
> rate?
>
> What I mean by this is, if I have an IIR filter F0 with a particular
> frequency response at sample rate S0, and I want to create a filter F1 to
> give, as near as possible, the same frequency response, at a different
> sample rate S1, is there a standard way of calculating the coefficients of
> F1 from F0?
>
> My initial thoughts are that maybe I can find the zeroes and poles, and
> then rotate them around the unit circle or something, but I suspect there
> may be a standard solution to this problem?
>
> Thanks for any help.
>
> Paul ...
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