Yes, you're absolutely right about this, Tim. I chose to do it this
way on my DM642 project since I don't yet know the architecture
of this machine very well and, being a parallel machine it's a lot
more difficult to pick up than, e.g., the TMS32054x. The routine
is also invoked at a very low ra...
Tim Wescott wrote:
> Jerry Avins wrote:
> > jim wrote:
> > > Tim Wescott wrote:
> > >
> > >
> > >
> > > > In this case, what happens in practices is that the unit-circle section
> > > > puts out a transient that will always integrate to the true DC
> > > > value,...
I have constructed a 256 pt complex fft. My scaling is causing
problems as the outputs from each stage are divided by 4 to avoid
overflow. The input data and the twiddle factor coeffs are 16 bits
wide (Q15). My problem is that when the data arrives at the last two
butterfly stages of t...
I'm looking for good links to information on Fixed Point Arithmetic.
The only good resource I could find was -
Fixed Point Arithmetic: An Introduction - by Randy Yates
But, it is only an intro and does not deal w...
My application is going to need a bit of fixed point, a bit of floating
Any advice as to whether a 720MHz fixed point processor (TI C64x) executing
floating point libraries, can compete with a 225MHz floating point processor
(TI C67x) ?
I searched the internet to find out the difference between a fixed
point & a floating point DSP.
To my surprise, I did not find a well defined difference.
Plz, can anyone differentiate between the two so that a layman can
also understand ?
Thanx in advance...
The 32 bit data types of the Blackfin processors should preserve more bits
of precision than a 16 bit fixed point processor, if the FFT routines make
use of the 32 bit data types. Can anyone comment on whether this is the
case, and how well the precision is maintained in the calculations? How
I think you must appreciate our own TTS/ASR engine, on
Becose 2 weeks ago I have finished SR for SpeechBox device, processor
TMS320VC5402 (fixed point). Sources are ANSY C 99%, dsplib-TI for
There isn't big problem to release for ADSP
On Thu, 17 Jul 2003 15:48:03 +0200, Andre wrote:
> I am looking for a simple function to get an phase angle from real an
> imag part preferable without dealing with atan and special cases.
> I do not need to be too accurate (1% is OK) but I am tight in code space.
Mr. Thomas already posted...
> I am starting to learn the DSP by using the Analog DSP. I am wondering
> which one is "best" - I do not have target application now. just for
> learning DSP. What I mean "best" is that the chip architecture is
> current, stable, widely used, easy use(?), and the support for the...
I am trying to implement an FIR and an IIR filter in fixed point
The actual filter is of no importance(so I have implemented the
simplest filter possible),as long as it is in fixed point,and moreover
as I can give as input the desirable precision.
I wrote the following piece...
I am Gowtham, i am doing project on DSP in speech compression, i have
written a program in C and i need to implement this on a fixed point
chip,so now i have to convert the floating point numbers into fixed
point and then implement the fixed point operations on the chip, i
would appreciate ...
Hi, I am trying to implement an IIR High pass filter to filter out a DC
component introduced from a ADC convter we are using.
I set the passband frq 200Hz, stopband freq 50Hz, Rp 0.2, and Rs -30dB.
Using Matlab, I got the following coefficients for using 2 2nd order biquid
I'm a relatively newbie to DSP and would like to add a resonant lowpass
filter to a simple sound application. I have a floating point routine to
do this but am wondering if this is possible in fixed point. I can't
easily convert my existing floating point routine as this involv...
> The simplest I know is:
> if(signal too big)decrease gain,
> else increase gain
That's that I already know. Your algorithm is about to change gain too
often, there should be some highmark and lowmark levels. The problem
is to determine is the signal is big or small.
> Can't offer ...
robert bristow-johnson wrote in message news: ...
> In article email@example.com, Matt Boytim at
> firstname.lastname@example.org wrote on 07/30/2003 21:26:
> > A hardware multiplier is mighty powerful so a technique that utilzes
> > the multiplier, such as polyn...
email@example.com (Andrew Xiang) writes:
> In the matlab filter design filter box, when I click the Quantization
> button after design a IIR, the freq response seems very different. I
> cannot find the place where you can configure how many bits for the
> Where do you specifi...
Some more info:
Mike Rosing wrote in message news: ...
> yj wrote:
> > Hi guys,
> > I plan to learn programming TM320c DSP using C language. Is there any
> > book or manual for that?
> > Thanks
> > Jeff
> Tons. check the TI web site for starters and pick a proce...
I'm a newcomer to DSP and am trying to write a Fixed Point
implementation of the bilinear transform lowpass filter as outlined in
the Audio EQ Cookbook
y[n] = (b0/a0)*x[n] + (b1/a0)*x[n-1] + (b...
firstname.lastname@example.org (Theo van der Merwe) wrote in message news: ...
> email@example.com (Mahesh Godavarti) wrote in message news: ...
> > firstname.lastname@example.org (santosh nath) wrote in message news: ...
> > >
> > > I am sorry to reject the second job-meaning the second stat...
> I'd really like to get hold of the TI C26-based DSK (TMDS3200026),
> because it is (was) cheap & looks pretty simple. But, no-one over here
> (England) seems to stock them any more. The only supplier I've found
> won't sell me less than 5!
The DSP you mentioned is obsolete. I recommend y...
So I'm lookig for a DSP with the following characteristics :
- about 80-100 MIPS
- 16 bits fixed point operations
- low-power for embedded application
- on-chip flash (or EEPROM)
- at least 8 12-bits A/D's
As you see, my application doesn't require too much calculation pow...
email@example.com (Rune Allnor) wrote in message news: ...
> Jerry Avins wrote in message news: ...
> > There is no force in that example. It's just a starting point. Extend
> > the example ny imagining a railroad track running down the meridian, so
> > the moving object has to rem...
robert bristow-johnson wrote:
> Erik de Castro Lopo wrote in message news: ...
> > I'm therefore asking people here the following questions:
> > 0) How big and how "difficult" are the IIR filters you
> > have designed?
> i have chained together mayb...
"riya" wrote in message
> hello guys,
> I need some help from you. I am doing a DSP project and for that I need
> to do some C coding for the conversion of sample data which is in
> floating point representation to fixed po...
Peter H wrote:
> I have been looking a bit on this scrambling method and tried to write the
> source code for it.
> As I understand the input is a sample value and the numbers from 1 to M-1
"Jon Harris" wrote in message
> "Glen Herrmannsfeldt" wrote in message
> > "Jon Harris" wrote in message
> > > > > 1. Have you seen this form of oscillator before? Does it go by
"Anders Buvarp" wrote in message
> Hello Jim,
> Thanks for your reply.
> It is 0.5^x I need, maybe I can do a look-up table for now.
Well, 0.5**x (Fortran notation), is just a right shift if x is an integer,
and 0.5 is a fixed point number w...
I have to implement the FFT for a vector long 2048 samples in a DSP which
does not support floating point.
Do you have any link to already written code that computes the FFT with
Thank you in advance
Randy Yates wrote:
> There can be good reasons to AGC even in a digital discriminator, e.g., if the
> arctan() function is performed via a lookup table.
CORDIC has problems for very small magnitudes as well.
> Here's another example of a signal that is clearly bandlimited but
I am looking for an efficient fixed point FFT in C. Does someone know an
implementation on PIC18 (I know that PIC18 is not a DSP but this
question is in the frame of a feasibility study with high constraints on
You might want to look at a technical report on the web:
Generating Gaussian Pseudo-Random Deviates
It examines the accuracy of the sum of 12 uniforms approach and suggests a
new algorithm that should be easy to imple...
In a nutshell you divide the integer value of (e.g.) 8192 by 2^b,
where b is the "scaling" of the fixed-point value. In your case, b
must be 13 since 2^13 = 8192 and 8192/8192 = 1.0.
In the second example, the scaling of the value is still apparently
13, since -16385/2^13 = -2.00012207,...
Jim Frohoff wrote:
> > I believe he means that the white noise only needs to be around the
> > frequency he is trying to mask.
> > > So I believe the answer is that you need to filter the white noise
> source to
> > a tunable bandwidth. (Filtered ...
On Sat, 11 Mar 2006 18:00:44 +0100, Roman Rumian wrote:
> Hello Friends,
> what is the advantage of using fixed point numbers in DSP calculations ?
> Could anybody explain, why digital signal processors use it instead of
> integers ?
It's mostly a convenience, so that the most sig...
I am running an 1 tap complex tracker with an adaptive weight using a
The floating point algorithm works well with this value of ...
"Al Clark" a écrit >
| To calculate the SPL (sound pressure level), you need to know the
| sensitivity of the transducer (mic?) and the gain structure of the
| soundcard. With a typical sound card this is sometimes more
| than you think. The easiest way is to measure the output...
FFT libraries don't need to be that complicated.
A mixed-radix Fast Fourier Transform based up on the principle,
"Keep It Simple, Stupid."
There are many great fft libraries already around. Kiss FFT is not trying
to be better than any...