I had really just started to look at the software. I was using the default settings which has upsampling deactivated on the computer side and on the dac input section “For Bridge devices: Maximum Sample rate” set to "no limit.
The screen shows that my non-HD tracks start out at ALAC 16/ 44.1 khz and are converted to PCM 64/44.1 khz. But on the Dac input it is listed as 24/ 768 khz. (Which seems like upsampling to me even though upsampling is turned off.)
In any case, those settings work as long as the file is 16/ 44.1khz. However, as soon as I hit an HD track in my playlist, 24/96 khz. The play speed slows down. In this case, on the computer side it says ALAC 24/96 to PCM 64/96 to PCM 24/768.
What I found is that If I set the maximum sample rate on the DAC input to 96 khz, the file will play at speed and the audio output is alac 24/96 to PCM 64/96 to PCM 24/48 on the DAC input side. Which seems like it is downsampling.
I really don’t understand how these numbers fit together. Is there a tutorial in the software?