Milky

I’ve done some more research on this on the VAC website. In one section, it says that low numbers should be avoided, but elsewhere it says “The more is event frequency, the shorter is event period, the less is block size, the smoother is stream transfer, and the less is latency. But decreasing interrupt/event period causes increase of system timer resolution and timer interrupt frequency so system overhead is increased too. VAC driver sets system timer resolution to a half of the “MS per int” parameter specified.

Large (more than 10-15 ms) values may significantly disrupt stream uniformity and even cause stream breaking.”

I guess if you have a “grunty” PC, low (1 to 5) will work OK, but throw a greater load on the OS. Higher (10-15) may reduce the OS load, but could let too many other events get in the way of a smooth stream.

Glad to hear that you have made some progress. WHat happens if you just run the BAE wizard now and let it figure out the best buffer size? Sometimes, you can get too hung up on trying to manually tweak the parameters. So long as you can’t hear any artefacts like crackling in the background, a bit of jitter is normal. If you go overboard with large buffers, you will have problems with lip-synching with your karaoke videos because BAE will fill the buffers before doing any processing.

My son and I both have identical laptops, yet his settings are slightly different from mine. I assume its a firmware or chip revision thing, but I’ve never really looked into it.

Sometimes, to uninstall the Pipeline, you need to go into Device Manager and remove the actual Pipeline (after you have run the regular uninstall). It seems to have a service which needs to be unloaded manually.

I doubt that it is EVER possible to get 0 delay when the audio has to be processed through ASIO plus Breakaway, plus any overheads introduced in the MPX process. If your STL distance is significant (like 5 kilometres), there will be a noticable delay if you listen off-desk compared to off-air.

Obviously, using another device to give you some more control over the levels (a bit like using a mixer channel or Distribution Amplifier). Many radio stations have a DA rack in the control room to tweak all of the sources to a similar level. However, it intoduces yet another source of noise, distortion and breakdown failure risk.

Just to eliminate the Behringer, can you route BAE to your internal sound or some other device?

What happens if you ignore the Wizard’s settings and play around with the buffer settings manually? Usually, a buffer size in multiples of the sample rate works best (as in your 882), but sometimes, if the sound card has unusual frame geometry, a different buffer size works better. Start at 441 and run the test for each setting after that.

I am of the vintage that listens predominantly to the music you are playing. I do listen to SOME “modern” music, but much more ove the pre-Y2K period. These were times before a lot of compression and other processing, so you really don’t want anything too heavy.

My marketplace is not in a situation where I have to out-shout other stations much, so I can concentrate on good, clean, open sound without having to crash through other louder sources.

Personally, I like to start with the “Refernce Settings” preset and tweak the minor settings through the “enhance” option. This will level all the quieter 60s, 70s tracks with the louder 80s to noughties, without any over-driving.
If you want something a little more aggressive, try French Kiss or Quintessence.