I spent a good month every evening trying numerous ways to do make this work and came up with this. "This" is basically my understanding and if someone else has come up with something better I'd be really interested in hearing it.
The AxeFXII has to be the
master if it is the audio source (which is probably about 99% of the cases) when using digital. It does not have a dedicated word clock input. The only time you would set it as slave is when re-amping or using the digital input for whatever reason. I'm sure you know this, but for others it is basically that wherever the digital signal originates that clock has to be used and wherever it is going has to be locked to it since there is no way to sync it any other way.
Its really cut and dry when you think about it that way. And that's not an AxeFXII only issue, any device that has no word clock input will act that way.
The downside is that if you are re-amping you have to change the master clock source after you initially record and when you re-amp. Many DAW's won't let you do that because the sound card is in use. Some apps will allow you to 'release audio driver in background' so just clicking outside of the DAW window and then opening up the sound card's control panel will let you change the master clock setting. The other downside is that once the AxeFXII is turned off and you forget to set the master clock source back to the sound card you get no sound.
This applies to AES as well.
As for the 48kHz thing, it's been beaten to death on here over the years. It appears to me that it comes down to Cliff not wanting to use an internal SRC because of the impact on quality of the sound. I"ve heard this with the 11R where everything except for 96kHz sounds like mud (probably uses 96kHz internally and then shoves it through an SRC onboard). It was very convenient to have multiple sample rates available, but not at the expense of fidelity. Conversely I've used the analog outputs of the AxeFXII into my sound card at various sample rates and there is no difference in quality because there was no sample rate conversion involved, just the A>D conversion itself I am assuming.
The only other real option I could find for re-amping capabilities at various sample rates without changing clock settings all the time was a
Mutec MC6 which is supposed to do bidirectional SRC and basically reclocks everything for you.
If you aren't planning to re-amp all the time then a standalone SRC that can synchronize to an external word clock
should work. I'm just skeptical about the sync issues.
To be perfectly frank if you have to use a sample rate different than 48kHz you are best served to use the balanced analog outputs into your sound card/audio interface. No pops, no jitter, no dropout and no drift and any sample rate your interface is capable of. Output 1 is the only truly balanced output so I'd use that. And the quality of the analog output is phenomenal.
