FM3 thru USB frustrations

On MacOS with multiple interfaces open AudioMidi Setup set Format for all audio devices to 24bit 48khz.

If you are using SPDIF out of the FM3 to a interface with SPDIF change the clock source of that interface to SPDIF(if it allows), The FM3 will be the master clock as there is no option to change that on the FM3. Using a hardware sync such as the SPDIF from the FM3 is going to be the preferred way to sync the audio clock with other devices. Other devices may have Word Clock, AES or Optical. If there is no hardware option then drift correction is the way to go.

Make sure your DAW audio settings are also 48k.

For aggregate device setups search for “set aggregate device settings in Audio MIDI Setup on Mac”


Also
 
Went back to aggregate device, made FM3 clock source.

question on Sample Rate: I have both devices set to 48k. However I see that the FM3 is 48k 4-channel 24 bit integer. When I click on the Apogee Element 24, it is 48k 16 channel 32 bit integer. Does it matter that the one is 24 bit and the other is 32 bit?

thanks!
 
Luth2000 said:
Ok - went 100% FM3, input and output, output 1 to speakers. Initially still had artifacts, but THEN disengaged input monitoring in Logic, and no artifacts!

Interesting - tried some other settings in logic having to do with "Audio sync mode" in "Project settings". default is MTC Continuous. I tried "external or free". There is another option that is "MTC trigger + autospeed detection" - I wonder if that's what I should be trying.

Anyway, seems like problem solved! Well not really for my purposes (but am glad to see some clarity on the artifacts). I use input monitoring, for I like to use plug-in effects on my guitar sound. When I run thru FM3, I lose the ability to control input monitoring, as well as set levels within Logic to balance the guitar tracks "pre-recording". I've always tried to avoid setting those levels on the fractal side. I suppose I can change my ways on the leveling of tracks, but how do I keep my plug in use? Sure I can add plug-ins after the fact but then I've no nuancing the guitar sound I want when recording. I'll do some web searching, but this issue is why I bough the Apogee Element - so I can monitor outside of the fractal world.
Click to expand...

That should work fine. I'm not sure what isn't working for you at this point, and I don't understand why you're using the Apogee, but MTC has nothing to do with this. As I mentioned above, your problem is clock drift.
 
Luth2000 said:
Went back to aggregate device, made FM3 clock source.

question on Sample Rate: I have both devices set to 48k. However I see that the FM3 is 48k 4-channel 24 bit integer. When I click on the Apogee Element 24, it is 48k 16 channel 32 bit integer. Does it matter that the one is 24 bit and the other is 32 bit?

thanks!
Click to expand...
Bit depth doesn’t matter for clocking only sample rate. Bit depth 16/24/32 etc. is all about signal to noise(SNR). Many devices and DAW’s use 32bit float which is 24bit with 8 additional bits for headroom/dynamic range.
 
Here’s another update. I thought I had it all figured out by creating an aggregate device, and making the fm3 the master clock, but then after a while crackling came back. Ugh. So I turned to apogee resources, and discovered that many of the apogee owners community have inherited this problem in the last 1.5 years due to changes in macOS. Apparently apogee made efforts to rectify, but some forums suggest they have given up in the last 6 months...

Glenn asks why I want to use another interface. Here are my reasons:
1) I want to use plug-in effects when recording my guitar tracks. i have some killer Valhalla Reverb’s and delays I really like, as well as others.
2) the sound quality of my non-guitar tracks is superior coming out of the apogee than the fractal. for example, the piano out of keyscape is definitely more detailed and higher fidelity coming out of the apogee.
3) I prefer to control input levels through logic.
4) I want to run all software instruments, an additional axe fx 2, a usb mic, etc all through one audio interface, and prefer to use the fractal for what it’s best at - delivering awesome guitar tones.

those are the reasons why, but today I am running through the fm3 only.

Im not too optimistic the apogee is going to be resolved, so I may be in the market for a non-apogee replacement. If any readers are still around and can identify with what I’m shooting for, and have a mac system that is working 100%, I’d be interested to hear which audio interface you are using. I’m running Catalina.

thanks for the support everyone!
Luther
 
Luth2000 said:
Went back to aggregate device, made FM3 clock source.

question on Sample Rate: I have both devices set to 48k. However I see that the FM3 is 48k 4-channel 24 bit integer. When I click on the Apogee Element 24, it is 48k 16 channel 32 bit integer. Does it matter that the one is 24 bit and the other is 32 bit?

thanks!
Click to expand...

What sample rate and bit depth are you recording to? Ideally, your interface settings and project settings should match to avoid any unnecessary on-the-fly resampling and associated CPU load. Fractal gear is fixed at 48 kHz, 24 bit, so if possible, set your Apogee and your DAW projects to that as well.
 
mr_fender said:
What sample rate and bit depth are you recording to? Ideally, your interface settings and project settings should match to avoid any unnecessary on-the-fly resampling and associated CPU load. Fractal gear is fixed at 48 kHz, 24 bit, so if possible, set your Apogee and your DAW projects to that as well.
Click to expand...
Yes everything 48. The bit rate of the apogee is 48 32 bit rate, but someone on this thread told me that isn’t a big deal.
 
ksandin said:
On MacOS with multiple interfaces open AudioMidi Setup set Format for all audio devices to 24bit 48khz.

If you are using SPDIF out of the FM3 to a interface with SPDIF change the clock source of that interface to SPDIF(if it allows), The FM3 will be the master clock as there is no option to change that on the FM3. Using a hardware sync such as the SPDIF from the FM3 is going to be the preferred way to sync the audio clock with other devices. Other devices may have Word Clock, AES or Optical. If there is no hardware option then drift correction is the way to go.

Make sure your DAW audio settings are also 48k.

For aggregate device setups search for “set aggregate device settings in Audio MIDI Setup on Mac”


Also
Click to expand...
Ok - educate me on SPDIF. Do you use that INSTEAD of USB, or in addition to? Does SPDIF transfer the data needed to run axe edit? Can I run without USB?

thanks!
 
Bit depth is easier to convert than than sample rate, but it still adds unnecessary processing that gives no real sonic benefit in this case. Extra bit depth for processing is always helpful for added accuracy, but since this is just unprocessed input from the interface, you might as well set everything the same.

SPDIF is just a stereo digital audio stream. It caries no other data. USB is needed for FM3 edit, etc. If your Apogee interface as a SPDIF input, you can send your audio stream from the FM3 to it that way as well. It is also 48 kHz, 24 bit audio, that is identical to the signal coming through USB. Keep in mind that SPDIF is only 2 channels though, while USB is 8 channels (4 in and 4 out). The FM3 also does not have SPDIF input like the Axe III so it would only be useful for output to another interface.

If you are monitoring through your Apogee, SPDIF does have the benefit of no additional latency as long as the Apogee allows you to direct monitor the SPIDF input stream.
 
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mr_fender said:
Bit depth is easier to convert than than sample rate, but it still adds unnecessary processing that gives no real sonic benefit in this case. Extra bit depth for processing is always helpful for added accuracy, but since this is just unprocessed input from the interface, you might as well set everything the same.

SPDIF is just a stereo digital audio stream. It caries no other data. USB is needed for FM3 edit, etc. If your Apogee interface as a SPDIF input, you can send your audio stream from the FM3 to it that way as well. It is also 48 kHz, 24 bit audio, that is identical to the signal coming through USB. Keep in mind that SPDIF is only 2 channels though, while USB is 8 channels (4 in and 4 out). The FM3 also does not have SPDIF input like the Axe III so it would only be useful for output to another interface.
Click to expand...
Would hooking up SPDIF improve clock syncing over USB?
 
mr_fender said:
Bit depth is easier to convert than than sample rate, but it still adds unnecessary processing that gives no real sonic benefit in this case. Extra bit depth for processing is always helpful for added accuracy, but since this is just unprocessed input from the interface, you might as well set everything the same.

SPDIF is just a stereo digital audio stream. It caries no other data. USB is needed for FM3 edit, etc. If your Apogee interface as a SPDIF input, you can send your audio stream from the FM3 to it that way as well. It is also 48 kHz, 24 bit audio, that is identical to the signal coming through USB. Keep in mind that SPDIF is only 2 channels though, while USB is 8 channels (4 in and 4 out). The FM3 also does not have SPDIF input like the Axe III so it would only be useful for output to another interface.
Click to expand...
I don’t see a way to change the bit depth on the apogee. I have all sorts of rate options, but no depth options.
 
Online specs for the Apogee show its max sample rate is 192 kHz, 24 bit. The manual also makes absolutely no mention of bit depth anywhere, even in the specifications. Quite odd.
 
My Apogee Duet is 24 bit. There's really no reason these days for an audio interface to support anything else. 32 bit float would be nice, but that's not happening yet. Anyway, like you said, unlike sample rate, there's no need to worry about bit depth conversions.
 
Luth2000 said:
That was a flawed question. Better version: would hooking up SPDIF for clock syncing give better results that syncing over USB?
Click to expand...

No, clocking is not inherently better over SPDIF than USB. It's almost always better to use USB unless there is a reason why you must use SPDIF.
 
I don’t have as much faith in relying on USB for Audio sync. I’m old school, in some ways, prefer hard wired. YMMV.

“If any readers are still around and can identify with what I’m shooting for, and have a mac system that is working 100%, I’d be interested to hear which audio interface you are using. I’m running Catalina.”

I’m running 2015 iMac, Catalina, Logic with Arturia Audiofuse Studio interface. No issues. The Audiofuse Studio functions without the iMac, which was one of my gripes with other choices.
 
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