"BitRadio" by Thomas Rode/DL1DUZ - First steps: Before starting the program, make sure the audio-cable is plugged in. At some PCs Java expect the cable to be in when starting up. Otherwise the underlying hardware (audio-port) will not be recognized. If everything was done correctly, "BitRadio" should open a window and a ticker-like display at the bottom will inform you on the settings missing and actions to be performed. At first use, the following has to be set: * Under "Sound" (top left) an input line and output line has to be selected. That can be either a port (actual interface at your PCs soundcard, like Microphone, Speaker, ...) or a software mixer. In most cases the port "Line In" will be used as input. Sometimes, if the signal coming from the converter turns out to be too weak, "Microphone" is the better choice, since on most PC there is an extra amplifier built in for that input. Finding the most suitable setup may require some experimenting but fortunately has to be done only once. If a software mixer is set as input (and/or output), any other program that can interface that mixer can become a source (destination) of the signal. That feature is particularly valuable if you want to experiment with BitRadio and maybe even write your own programming code around it (see release note 0.5). To enable that, all Volume Control, which is normally done by BitRadio when using a port, will be turned off. Control has to be achived either by hand or by external software. Remark: Make sure that the soundcard/mixer has all sound-processing (noise-suppression, echo-cancelation, equalizing, ...) turned off. Any modification of the signal prior to the actual SDR will lead to (severe) distortion. * Once you have set the line, the software will search for the respective mixer and list all available sampling modes. The results can be found under "Sound" / "Input Mixer" and "Output Mixer". Select the right mixer/mode. The software supports input mixer for 48, 96 and 192kHz (if hardware permits). The faster the sampling, the more bandwidth will be available on your radio. On the other hand computational cost increases too. Although "BitRadio" is optimized to make best use of the resources available, some older PCs may not be able to run at 192kHz. Remark: At Java, linking the port to the mixer is somewhat fuzzy. So I had to use a mathematical function called "Cross-correlation" to "guess" which of the mixers should be right. Being a smart human, please support my algorithm by double-checking the list it generates (naming etc.). * When the mixers have been set, you can select the right input- and output- volume control. The output volume control simply transfers the control of the audio volume to the two little +/- buttons at the SDR-window. That allows you to control the volume without having to open an extra window at your PC. Remember, that function is off when the output goes to a software mixer. The level of input volume is visible to the user but, besides that, fully controlled by the software to adjust the "gain" of the receiver depending on the current signal. Remember, that function is off when the input comes from a software mixer. The software will offer you all possible volume controls available for the respective mixer. Here again you have to use your brain-power to help the simple software. In most cases something like "Master Volume" will be the right choice. * Having reached this point (and provided that the software feels happy with what you selected) the biggest chunk of work lays behind you. The software should have started sampling, you should hear some sound and the waterfall should start to run. What remains to be done is: - setting the center frequency (the "beat" of your converters local oscillator) under "SDR". - selecting the CW side tone frequency (that is the tone a signal at Zero frequency makes when receiving in CW-mode). - pre-setting the cutoff frequencies of your filter banks. - setting/selecting the desired FM-deemphasis when using FM-mode (that feature was added in release 0.3/see release notes). Good news is that all settings made are being stored in a file "SDR_settings.ini" and reloaded at the next start of the software. Sometimes (if something goes wrong) some incorrect data can find its way into that file, preventing the software to correctly initialize next time. In this case the only way out is to delete "SDR_settings.ini" and start from scratch. This happened to me only during testing of the software and when I was being overly "creative". In other words, this should not happen to you at all. Finally it should be mentioned that all settings can be changed at any time. Some, though, may require restarting the software afterwards (the line at the bottom will tell you). - What else can be found on the SDR-window: * waterfall-display (it is about 30s long) / For more information on that item please refer to the WWW. * DC correction / At the center of the waterfall (center frequency) the DC-part of the spectra (Zero frequency) can be found. A "real" DC-component coming from your system (converter, soundcard, ...), as well as all input-signals on the center frequency will be converted to here. The first should be eliminated. To achieve cancelation of the "real" DC the software estimates and removes that portion of the signal. The unchanged signal is, however, displayed at the waterfall diagram, allowing the user to evaluate its hardware. So if you notice a conspicuous line at the center position, some of your hardware generates a DC-offset. Since that reduces the dynamic range too (removal inside the software only happens in the frequency- domain), some measures to eliminate the DC should be taken (if possible). * frequency-needle / If the mouse pointer is moved over the waterfall, the needle turns from Red to Blue, indicating dial- mode. You can change frequency by either pressing a mouse button (scan-mode) or by double-clicking a mouse button (step- change). * filter-settings / There are two lines indicating the low and high cut-off frequency of the built-in filter. You can change those similar to the receiving frequency by moving the mouse pointer over the respective silder (Light Blue line) and double-clicking or pressing a mouse button. * receiving mode / Selects the current receiving mode. / SSB-USB, SSB-LSB, CW, CW-N(narrow), AM, FM, OFF(no demodulation; the complex output signal is directly passed to the two audio channels for further processing) * input volume / Displays the volume set by the software to control the receivers input gain. For optimal performance the number should be in the range of 40...80%. If the level is at 100%, your converters signal is too weak and you are loosing dynamic range. Numbers below 40% indicate a too strong input leaving the software little control-margin in case of changing receiving conditions. If a software mixer is choosen as input, the software still calculates this value but no adjustment is made at the mixer. * output volume / Displays the output volume set by the user. If a software mixer is choosen as output, the software still displays this value but no adjustment is made at the mixer. * signal meter (s-meter) / For more information on that item please refer to the WWW. * vector diagram (vector display) / That provides you with information on the relative angle and amplitude of the I- and Q- signal. If you do not want to look behind the curtain, all you have to be concerned of is the two arrows being in the right position and having the correct length. The Q-hand belongs in the 12 o'clock position, whereas the I-hand is fixed at 3. Both hands should have there tips ending at the circle. If both conditions are met, the tip of the hand will get a little green spot. Since the software performs an averaging over all representative frequencies, angle and lenght of the vectors can be subject to some minor changes. This "wobbling" is ok and does not result from issues at your hardware as long as it is not too big (+/-5°, +/-5% length as a rule of thumb).