How can I use the K2's resampling feature to convert an on-board song into an audio file?
If you have a K2500 or K2600/2661 with the sampling option, and enough Sample RAM, you can turn a Song into a Sample, by having the K2 resample itself while it is playing the song. This can be useful if you need a quick way to convert a song to a standard sample format such as WAV or AIFF. Many people want to do this so that they can create an audio CD with WAV files they have loaded onto their computer, without the need for other digital audio recording software or hardware. Following is a step by step description of how to do this:
Please note that although this tutorial goes through the process of sampling a song which is being played from the K2's own Song Mode, the basic process is also the same if you are using an external sequencer to send MIDI data to the K2.
If you have a K2500 without the KDFX option, or if you have a K2600 without the DIO-26 option, the Input parameter will be disabled once you set Source to Internal. This is because without those options the audio is always converted from Digital To Analog, and therefore the Input will be Analog. If you are planning on burning an audio CD with the WAV file you are creating, you will want to set the Rate to 44.1k, since an audio CD must be at that rate.
If you have a K2500 with the KDFX option, a K2600 with the DIO-26 option, or a K2661, then you two choices - you can set Input to Digital or Analog. To get the cleanest possible signal, you would want to choose Digital input. However, there is one disadvantage to this method - KDFX outputs only at 48kHz, and if you want to create an audio CD with your WAV files, you need it to be 44.1kHZ. Therefore, the sample would need to be converted from 48kHz to 44.1kHz after the sample is created. Once the sample is saved, you can convert it internally in the K2 using the Resample DSP function in the Sample Editor, or with external software on your computer. If you do the conversion internally, you should set the Quick parameter to 1 to get the best quality conversion. The chief disadvantage to doing it internally is that it can take a very long time for the K2 to complete the conversion, especially with longer samples. If you do the conversion externally on your computer, it will may be much quicker. But you should be sure the software that does the conversion does a high quality job, or the audio quality may be compromised.
If you have a K2500 w/KDFX or a K2600 and you choose Digital for Input, then the Cable parameter must be set to Coaxial. The display will show LOCK - 48kHz, showing the sample input is locking to the clock of the digital out. You will notice that if you switch the Cable to Optical, the display no longer shows that it is locked. The Out parameter on the K2600 should be set to Dir (Direct). The Format parameter is unimportant in this situation. Also, you will want to make sure the Digital Out parameter (Master/MAST2 page) is set to 16 bit (which is the default). Since the Kurzweil is a 16 bit sampler, setting the Digital Out to 20 bits will only result in those last four bits being ignored, which will cause low level audio artifacts. With Digital Out set to 16 bit, the signal will be properly dithered.
If you have a K2661 and you choose Digital for Input, the display will show LOCK - 48kHz, showing the sample input is locking to the clock of the digital out. The Format parameter is unimportant in this situation. Also, you will want to make sure the Digital Output Length parameter on the Master2 page is set to 16 bit. Since the Kurzweil is a 16 bit sampler, setting the Digital Output Length to 24 or 20 bits will only result in those last bits being ignored, which will cause low level audio artifacts. With Digital Output Length set to 16 bit, the signal will be properly dithered.
The other option altogether is to set the Input to Analog. Because the signal is now converted from digital to analog, and then back to digital, some noise may be added. But if you make sure to set the gain so the signal level in the loudest passages is as close to possible at 0dB, and if you don't have very quiet passages in your song, this noise may not be noticeable at all. The advantage to this method is that you can set the sample rate to 44.1kHz, so no rate conversion after sampling is required.
If Analog is chosen for Input, then the Monitor parameter is disabled, (there is no need to monitor the input since you can already listen to the output of the instrument. (In some older versions of the K2500, the Monitor parameter was not disabled, but if you have any early version of the OS, you should update your unit. )
Normalizing the Sample
If the maximum dB level shown in step 7 was too low, you will want to Normalize the sample. Press Edit (which brings you to the TRIM page), then DSP. Assuming you haven't been doing any other DSP Sample editing, the K2 will come up on the Normalize function. This function will increase the amplitude of the entire sample (assuming you don't change the Start or End points on this page), bringing the signal level up, just before the point at which the loudest part of the sample would clip. Press Go. For long samples this processing can take several minutes. Once it is finished, the K2 asks if you want to keep the change. Press Yes, then Replace. After a short wait (longer for lengthier songs), the display returns to the Normalize page. Press Done to return to the TRIM page.
Trimming the Sample
If you edited the End Point of your song so that it was long past the point at which audio had decayed away, or if you were pressing Stop manually to stop sampling, you may have an end point which is a number of seconds past where the audio has decayed below an audible level. Therefore, you may need to edit the End point of the sample so there is no "dead space" at the end of the sample, especially because this will make the size of the WAV file larger than it needs to be.
Here is a fast way to trim the sample:
Go the TRIM page in the sample editor if you are not already there. You will probably want the display to show Seconds instead of Number of Samples (the Quick Access button will switch between the two methods). Change the Start point to an amount a few seconds before the End point. Scroll over to the Loop Point and press Link (the Song button), then scroll to End and press Link again. This links the Loop and End points so that moving one also moves the other. The reason you want to do this is that all data is saved within the four points, so if you shorten the End but not the Loop, you won't end up making the file size any smaller.
Now strike your root key. The sample will play from the Start point you just chose until the End. If you don't hear anything, the start point may be past the point at which audio had decayed away. In that case, you can highlight Start again and move it to an earlier point. Now, with End highlighted, you can adjust the End point, striking the key and listening, until you have the End set just past the point at which the audio has decayed below an audible point. Once you have the End point set, don't forget to go back and set Start back to 0. Then press Exit and save the sample.
Changing the Sample Rate
As discussed in Step 3, if you have sampled with the Input set to Digital, the sample Rate is fixed at 48kHz. Since audio CDs need to have the sample rate at 44.1kHz, you would need to convert the sample rate. If you choose to do this in the K2, you would use the Resample DSP function. To do this, press the DSP button from within the Sample editor, then change the Function parameter to Resample. Set the New Rate parameter to 44100Hz and set Quick to 1. Then press Go. Please be aware that this process can take a very long time, especially with longer samples. Once it is finished, the K2 asks if you want to keep the change. Press Yes, then Replace. After a short wait (longer for lengthier songs), the display returns to the Resample page. Press Done to return to the TRIM page.
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