I have installed the latest version of Audacity on my iMac. I use OSX Mavericks as operating system. For sound recordings I use the external audio interface Edirol FA-66, which you have to connect on the computer by using a FireWire 400 cable.
(The audio-interface runs with Core Audio, which means you cannot install drivers on OSX.)
When I run Audacity and I use my audio interface as the output for sound I encounter problems: I hear clipping. The clipping is not in the sound-files. Also when I use the FA-66 as input the recordings are fine (which I can listen back through other applications like iTunes). The recordings made by Audacity sound even better than by Logic Pro X: that’s why I want to continue using Audacity. Also I am more familiar with Audacity than with Logic Pro X
I found out that the phenomenon ‚clipping’ occurs a lot less and sometimes it doesn’t occur at all when I run the application Jack Pilot together with Audacity. (I don’t know what the applications Jack or Jack Pilot are; someone gave me the advise to try it). But mostly the clipping phenomenon is still present.
I suspect the problem has something to do with the samplerate-output, though I make sure the desired samplerate (96 kHz) is selected the same under Audacity.
Sounds like your Edirol is getting the signal twice. Could it be Jack is routing the audio twice?
I don’t think this is a Firewire problem, cause that usually surfaces as in interfaces not working, clicks or dropouts. And I know the Edirol FA66 to work with a recent Mac with FW800 ports and Mavericks, as I’ve used one.
Please give us all three version numbers of Audacity from Audacity > About Audacity… , not just say it’s the “latest version”.
If the problem could be described as “clicking”, have you tried what would be the most obvious solution - reducing “Audio to buffer” in Audacity’s Recording Preferences? That setting affects playback too. The fact the problem is lessened with JACK also suggests that latency/buffering is the cause. Take the buffer setting down to 0 milliseconds, then if the audio does not play, increase buffer by 10 millisecond increments until it does play.
Thank you for your advise. I reduced the “Audio to buffer” from 100 milliseconds (standard) to 10 and now it works perfectly! (Setting it to zero doesn’t make it play audio)