Let me give it a try for ya. These instructions should work fine on a mac or a PC running Audacity. Probably Linux too, though I have never tried it.
1- Run the masters so the levels peak below -3.0 dB on each if you can, so you can mix them without having to lower volume to avoid clipping peaks. Run them both at the same sample rate. Bit depth is not as important to match. Name each file descriptively including the type of source (Foobar2008-04-29-SBD-24bit.wav, Foobar2008-04-29-DPA-24bit.wav or some such)
2- Go to AUDACITY>PREFERENCES> QUALITY and set your default sample rate and bit depth to the desired settings (44100 with the desired output bit-depth). PROJECT>IMPORT both sources into audacity and FILE>SAVE your work.
3- Use the double-headed arrow "TIME SHIFT" tool in the top left corner (<-->) to line up the files as near as possible to the start of the music. Find a sharp peaking impulse like a drum hit or some other peak to line up and zoom in until they are as precisely aligned as possible.
4- If you think you need to do any EQ or level changes on either source, do this now. You can use EFFECT>AMPLIFY to tell you how high each source peaks. If each source peaks higher than -3.0dB at any point, you will get clipping on the mix, so you'll need to lower levels to avoid this. It is also possible to have one source higher than the other and still clip peaks. I am good but not great at math, so hopefully some other folks can chime in with suggestions and comments regarding levels. I think I have it right, that 2 signals of -3.0dB will add up to peak at -0.0dB so you want to stay below that.
5- Go to the end of the files and figure out how much they have drifted apart. Do some math* to figure out how much you need to use EFFECT>CHANGE SPEED to get them lined up (see below for my method of calculating the percent change)
6- I suggest shortening the longer file rather than stretching the shorter one, but it probably doesn't matter. Use the EFFECT>CHANGE SPEED to do that.
7- Check the alignment to make sure the sources stay together. If they are not correct, use EDIT> UNDO SPEED CHANGE and try step 6 again. When they are correct, FILE>SAVE your work again. As long as you keep the file open, you can UNDO past the file save operation, but once you close and open it again, you can't go back past the saved version.
8- Check the mix for sound by using the MUTE function on each track during playback to make sure it sounds good. Adjust the gain for each track if needed by using the +......- slider on the left of each track for course adjustment, or EFFECT>AMPLIFY for finer control.
9- Go to AUDACITY>PREFERENCES > QUALITY > HIGH-QUALITY DITHER and select "Triangle Dither"
10- FILE>EXPORT AS WAV (or AIFF) to make the mixed file. FILE>SAVE again. If you think you might need to make further adjustments after checking the completed file, keep this project session open so you can UNDO back. Name the file something descriptive like Foobar2008-04-29-MIX-24bit or Foobar2008-04-29-MATRIX-16bit so you can distinguish it from each source file.
11- FILE>OPEN a new Audacity project document and PROJECT>IMPORT the newly created mix file. EFFECT>AMPLIFY to check that all peaks are below -0.0dB. If this plugin does not offer to boost levels, then you probably have clipped a peak somewhere, and you will want to go back to the original files and lower the levels of one or both sources to preserve your dynamics and avoid flattening out peaks. If you have a little headroom and it sounds good, then you have successfully mixed your sources.
12- If you want to track for CD's, then VIEW>SET SELECTION FORMAT > CDDA min:sec:frames 75fps and then EDIT>SNAP TO> SNAP ON to allow you to cut tracks without "sector boundary errors." Select tracks in order by using EDIT> MOVE CURSOR TO TRACK START (I go into preferences and give it a keyboard shortcut to make this easier) then shift-clicking on the end of each track, then EDIT> SPLIT each track apart in order, making sure to split the final track too.
13- FILE> EXPORT MULTIPLE (NUMBERING CONSECUTIVELY) to WAV (or aiff) in your selected target directory.
14- Compress these files losslessly using your favorite FLAC encoder, and upload to your favorite sharing website, and post in the KICKDOWNS thread here so we can check it out.
15- please let me know if this is unclear or can be improved upon.
* oh shoot, now I gotta figure out how to tell you the math part! My apologies for the half-assed nature of this part of my method.
Go to VIEW>SET SELECTION FORMAT > SAMPLES (SNAP TO SAMPLES) so you can measure the length of your program in samples. Measure the total length from your sync points early in the file to the desired sync points late in the file. You will get different numbers for each file since they are probably not lined up perfectly due to slight variances in the clock chips of the two recorders. Make a note of each of these numbers. Subtract one from the other to find out the number of samples of drift at the end, and write this number down. Divide the length of the longer one by the length of the shorter source and you will get a number close to but greater than 1.0000000. Let's use an example where you have exactly one second of drift at the end of exactly one hour at 48KHz. The longer file is now 172,848,000 samples and the shorter one is 172,800,000 samples. Divide the long one by the short one and you will get 1.0002788. (If I am getting this right, then) this tells you that you that you need to speed change the longer file by -.02788%
Damn I hope I got that right. Please won't someone troubleshoot my math and let me know the best way to do this???