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ADSP21371 - ADAU1361

analog@sgaur.hosted.jivesoftware.com — Fri, 20 Nov 2009 09:27:49 GMT

Hi to all

I want to connect audio codec ADAU1361 to ADSP21371. For configuring the codec i use the SPIB. In some example i saw that is used flag signal. Is this signal is for setting the CLATCH signal tree times low. And how this is achieved?

BAD LINK ON VISUALDSP++ RUN-TIME LIBRARY FOR SHARC

analog@sgaur.hosted.jivesoftware.com — Fri, 20 Nov 2009 20:07:57 GMT

On the 21469 Manuals page, the link for the last version of the run-time library points to the C compiler manual.

Please, review it.

German

Calculating Filter Coefficients to Store in a Microcontroller

analog@sgaur.hosted.jivesoftware.com — Fri, 20 Nov 2009 20:16:45 GMT

Recently there have been some email support requests regarding the calculation of filter coefficients (and other types of parameters) for storage in a microcontroller. The basic idea is to store a table of parameters in the microcontroller and download them to the appropriate addresses in the SigmaDSP's parameter RAM when required. SigmaStudio includes a few tools to help make this possible.

Method 1: Direct coefficient calculation

This method is not really preferred because it takes time and effort compared to the other methods, and doesn't offer any real benefits. In the help file, equations used for calculating first-order and second-order (biquad) IIR filter coefficients are given. For example, for a butterworth LPF:

These equations can be used to calculate parameters directly, but are a bit cumbersome.

Method 2: Fixed Point Filter Table Generator

Open the fixed point filter table generator under the Tools menu.

Here you can enter in parameters for a number of filter curves that you desire, then hit the generate button to see the generated curves on the graph.

Click the value table tab to see the biquad filter coefficients to be stored in RAM (5 coefficients are listed for each filter generated).

Click the Show Values button and then Generate again to show the decimal values and parameter names for each filter, along with the associated boost.

These values can be loaded directly into a 2nd-order general filter, set to IIR Coefficient mode in SigmaStudio. Important: Because of the way the values are stored in RAM, the coefficients for a1 and a2 must be inverted from what is shown in the filter table generator. The filter table generator shows coefficients as they are stored in RAM, whereas the IIR Coefficient filter box shows their true values as calculated by the equations.

The filter table generator allows a large number of coefficients to be calculated and stored in RAM for applications such as a user-controllable tone control.

Method 3:Use the capture window

Start a project and make sure the SigmaDSP is connected to a USBi communication channel in the Hardware Configuration tab. Insert any filter into your project and compile. Then change the filter values. The values written to the SigmaDSP's RAM are shown in the capture window (open by default, toggle by pressing Ctrl+5).

This allows you to simply generate the parameters for one filter curve.

Figuring out parameters for any SigmaStudio algorithm control using the Capture Window

You can insert any cell into the project, compile, then change its parameters to see what should be stored in RAM in the capture window. If, for example, I wanted to to a sine tone sweep that is manually controlled by a microcontroller, I could enter in a range of values into a sine tone generator and make note of the frequency parameter in the capture window (in this example, the parameter name is sin_lookupAlg19401increment).

how connect data dsp processor to PC?

analog@sgaur.hosted.jivesoftware.com — Fri, 20 Nov 2009 19:18:19 GMT

I make the sound with ADSP 21,369. I wanted to ask how to save the sound of that process into the file. wav on the computer? I use k-lite session to process voice and my 21369th ADSP sharc connect this with the usb port to the computer. how to save a sound process that results in the form of a file. wav for example? help to provide tutorial or program. thanks

Device serial number

analog@sgaur.hosted.jivesoftware.com — Fri, 20 Nov 2009 17:58:29 GMT

Is it possible to access the device serial number or any identifier unique to the device?

Thanks

Running exported makefile under Windows

analog@sgaur.hosted.jivesoftware.com — Thu, 19 Nov 2009 21:29:03 GMT

I can compile my code in VisualDSP++ but when I exported my makefile using "Project Options->Export Makefile..." and tried to run it under a Windows XP command prompt (i.e, cmd.exe) I get the following error:

set PATH=C:\Program Files\Analog Devices\VisualDSP 5.0;%PATH%

gmake-378 -f tmm_export.mak tmm_Debug

"Linking..."

'C:/Program\' is not recognized as an internal or external command,operable program or batch file.
gmake-378: *** [Debug/tmm.dxe] Error 1

This would appear to be due to the fact that I am running under Windows instead of UNIX because the tmm_export.mak file replaces

ADI_DSP=C:\Program Files\Analog Devices\VisualDSP 5.0

with

VDSP=C:Program\ Files/Analog\ Devices/VisualDSP\ 5.0

(replacing backslashes with slashes and spaces with backslash-space) which is what I assume is causing cmd.exe to choke.

Are there any environment variables or arguments to cmd.exe that I need to set to get gmake to run from cmd.exe? Can someone point me to the documentation which describes how to do this? How does the IDDE run the tmm.mak file properly?

I am using VisualDSP++ 5.0.6.0, IDDE 8.0.6.76 (Feb 12, 2009) under Windows XP Pro service pack 3.

system services dma callback

analog@sgaur.hosted.jivesoftware.com — Wed, 23 Sep 2009 07:02:37 GMT

hello,

I am using the system services and driver model to write a driver for the AD5663 DAC. I am building it upon the ADI sport driver with DMA using a two dimensional "ping pong" buffer. The idea is to fill one subbuffer, while the DMA transfere the other. This works fine, but I see a potensial problem in the way I get DMA events in my interrupt callback. The only event i get is the ADI_DEV_EVENT_SUB_BUFFER_PROCESSED. I never see the ADI_DEV_EVENT_BUFFER_PROCESSED. I had hoped to be able to determine which subbuffer the DMA had completed this way. My next approch was to use the pointer to the buffer received in the callback, but this pointer allways points to the start of the whole buffer, so there is no indication of what buffer that is actually processed. By examining system services source (adi_dma.c) I see the the following statements which I belive confirms my observation:

// CASE (circular)
case ADI_DMA_MODE_CIRCULAR:

            // determine if it's an inner loop interrupt or an outer loop interrupt
            if (pChannel->Config->b_DI_SEL == ADI_DMA_DI_SEL_OUTER_LOOP) {
                Event = ADI_DMA_EVENT_OUTER_LOOP_PROCESSED;
            } else {
                Event = ADI_DMA_EVENT_INNER_LOOP_PROCESSED;
            }

            // override the pArg callback parameter to be the buffer start address
            pArg = pChannel->StartAddr;
            break;

Should i use another DMA model in my case? Is there a way to be absolutely sure that my program is syncronized to the DMA buffer handling? When I stop the transfere, will the DMA be re-initialised to start with a clean buffer structure or may it start in the subbuffer?

mads-l

Release vs Debug

analog@sgaur.hosted.jivesoftware.com — Wed, 18 Nov 2009 19:15:49 GMT

Is there a global define provided by the IDDE to identify Release and Debug modes? I need to configure the PLL/SDRAM init differently for stand-alone vs emulator modes.

optimized goertzel algo for Blackfins

analog@sgaur.hosted.jivesoftware.com — Fri, 14 Aug 2009 06:17:28 GMT

hi !

is there any optimized code for a goertzel algorithm which is usable for blackfins?

regards chris

Bootloading Blackfin BF537 with VisualDSP

analog@sgaur.hosted.jivesoftware.com — Thu, 19 Nov 2009 16:15:39 GMT

Hi!

I'm starting to work with DSPs. I have a EZ-KIT with the BF537 on it. I can debug it using the integrated tools, no problem. My problem is, that I should use the Bootloader to get the code on it. I tried to change the preferences of the Project, but I have problems to bootload the code.

Can anyone give me some hints how to do it? Everytime I click on "Load" he want's to connect zu the integrated flash loader, and does not use the UART bootloader. What am I doing wrong?

Sorry for my bad english and many thanks for help.

Regards,

Andi

RMS results

analog@sgaur.hosted.jivesoftware.com — Thu, 19 Nov 2009 19:18:00 GMT

Hello,

I have recently received the ADAU1761 evaluation board. One thing I have noticed is that the RMS results using the RMS squared envelope and then the square root function work fine for frequencies above 50Hz. When I try an input frequency of 2Hz using the tone input, the results go up and we end up with a 30% error. I have found that if I can the TC of the envelope down to 5, reults are beeter with a 2% error. The problem with that scenario is the response time of the function. Changing the sampling rate did not affect the results either. Any suggestions will be appreciated.

Thanks.