Manual Online MM Ham. Soft. Version 1. 0. J June 1. 8, 2. JE3. HHT Makoto Mori. Translated into English by JA7. UDE ObaI have no plan to update this program any more. Please do not ask me any question about this program. This is a DSP filter tool using a PC with the soundcard. With this tool, you even can design various types of digital filters including adaptive filters. However, this tool is just experimental and will not afford the practical use for amateur ham radio. You probably need a powerful CPU to make this tool run flawlessly. In addition, you need a soundcard with the full duplex mode. I made this program just because of my own interest. As I was not quite familiar with the use of the soundcard, it still has substantial time lag from input to output and might not well work for CW. Needless to say, this program is freeware. How to uninstall DSPFIL does nothing to the Windows registry, so just delete all the files with the directory that has DSPFIL files. System requirement OS Windows 9. NT Note by VE5. KC has worked fine in tests with Windows XP VistaPC The faster the better. Desktop Author 7 0 1 Keygen Generator Mac. Display 6. 40 x 4. Soundcard 1. 6 bit soundcard that is capable of FULL DUPLEX some cards wont work Hookup and Operation Connect the speaker out of the radio to the Line in or Mic of the soundcard. Connect aheadphone or speaker to the output of the soundcard. Since the Mic input has too high gain, I recommend the Line in. Adjust the input level by using Mic or Line level in the Record property or the audio in the control panel. You can do that by using the AF gain of your radio, too. Adjust the output level by using Wave or Master level in the Play property or the audio in the control panel. You also can do that by using Up up arrow or Down down arrow button on the DSPFIL window. If you have a sound output from your speaker without running DSPFIL. EXE, your PC isconfigured so as to play the recording signal directly and thus you must turn it off. Go Playproperty and get Mic in or Line in muted. If you hear a sound immediately after starting DSPFIL. EXE, you are ready to go. In case you see a message like Cannot open the sound device, your soundcard probably does not support the full duplex mode. Give up listening to the filtered sound, but you can observe how DSPFIL. JbBl6k5xSgs/THUPW7TIfZI/AAAAAAAABUY/hyqeS15eb-4/s1600/keyboard+upperfilters.JPG' alt='Delete Upper Filter Lower Filter Vista' title='Delete Upper Filter Lower Filter Vista' />FITS File Image and Header Keyword integration with 32 and 64bit Windows Vista, 7, 8, 10 and beyond. Original Title Windows Explorer crashing consistantly Windows 7 Ive been getting a lot of error messages over the past few weeks regarding the Windows 7 Explorer. My kids went to change keyboard without turning off pc running vista home premium, with a ps2 connection. Now the keyboard doesnt work, or is not recognised by pc. EXE works by the FFT and adaptive filter response windows. Since there is a time lag between the input and output, you should keep the buffer size as small as possible. The time lag has a big trouble in filtering CW signals you will soon understand what it is when you transmit a signal, Hi. Too high input level causes distortion in the analog circuit of the soundcard. You have to adjust the input level by monitoring the FFT display set to IN. When overdriven, DSPFIL shows Over in the upper right corner of the FFT window. When the HPF button is depressed, the 1. Hz high pass filter is activated to the input circuit. It is effective if you have DC ingredient, but it raises the CPU load. Use it only when you need it. Details of the filters NS4NS1. This is a comb filter using moving average. This filter, by its structure, gets the actual centerfrequency Rfo shifted from the defined center frequency Fo by. RFo fss intfssFo Hz ifss fs 2j. This can be compensated by carefully choosing the sampling frequency fs. However, thesound blaster card does not allow fine tuning around 1. Hz, so DSPFIL admits the shift, Hi. The filter does not use a simple averaging calculation but uses subtractions for 12 periods. Thus, the even harmonics are suppressed, but the odd harmonics can be passed through. It is a good idea to use a 5. Hz filter of your radio. It has lower quality in the frequency domain compared with BP1. I think this filter gives the best performance particularly for weak signals. BP5. BP7. 0This is a band pass filter using an FIR filter. It uses x. 3 oversampling. The physical samplingfrequency is 1. Hz while the application sampling frequency is 3. Hz. If the number of taps is increased, the filter become sharper. However, it increases theprocessing time at the same time, and therefore it will not run on a slow PC. LMSBP, LMSB2This is an adaptive band pass filter for CW. I have not tested a lot on the values of mu and gamma, but I think the filter works, hi. This filter does not affect Fo or Tap, which is configured in the main window. The frequency domain graph in the lower right corner shows the frequency characteristics of the transversal filter calculated with the coefficients, which are changed by LMS. You can see how the adaptation is performed by changing the frequency of the input signal. In case of weak signals, the filter coefficients tend to be small, which would result in a low level output. To compensate this, LMSB2 leaves the AGC turned on to increase the volume for the weak signals. K1. 8. KThis is a fixed frequency BPF for SSB. The low cut frequency is fixed to 2. Hz. If itoversamples the 2. KHz or higher signals, it causes folding errors because of the decimeter. This filter does not affect the Fo, which is configured in the main window. LMSNSThis is a noise smoother for SSB. The adaptive operation might not be well tuned yet. The. SSB signal is smaller autocorrelation than the CW signals, so I put small values in the correlation delays. En La Gama De Los Grises. This filter does not affect the Fo or Tap, which is configured in the main window. LMSANThis is an automatic notch filter for SSB. It would give better results if it had faster convergence behaviour. However, I dare to focus on the response speed for CW signals. This filter does not affect the Fo or Tap, which is configured in the main window. User. User. 6This is a user customizable filter. The default setting gives a wide band pass filter for SSB. You can customize it by pushing the DESIGN button the button face text is written in Japanese. You can copy the parameters of the selected other filter to those of this filter by pushing COPY the button face text is written in Japanese button. This filter does not affect the Fo or Tap, which is configured in the main window. User setting for the adaptive filters. LMSBP, LMSNS, LMSAN are built in filters, but the user can design a LMS filter by himself. Push DESIGN this text is written in Japanese, so it might not correctly appear with non Japanese Windows button and select LMS, then push UPDATE in Japanese text button. Now you can change the parameters. The algorithm used in the adaptive filters is called Leaky LMSLeast Mean Square method. The user customizable parameters are Tap the number of orders of the transversal filter. Delay the number of delay nodes. V gamma the dumping factor. Larger u gives faster response but slower convergence. Smaller V gamma makes the coefficients decrease faster when the input signal is cut off. However, too small V gamma will result in oscillation. Generally put a value a little bit smaller than 1 to V gamma. If the REVERSE OUTPUT in Japanese is checked, DSPFIL outputs an error signal. It ischecked to design an automatic notch filter. When AGC is checked, DSPFIL automaticallyincrease the output volume for weak input signals. The characteristics of the adaptive filters are dependent not only on u mu and V gamma but also on Delay and Tap.