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This is probably dumb and/or already been asked, but will the Axe II load existing
AXE I 1024 bit IRs or do they have to be converted to a different format?
If they do need conversion, is it necessary to have the original .wav to do so?
The Axe-FX uses higher sampling rates (oversampling) during the processing stages. This is how it avoids aliasing when non-linearities are applied. But the sampling rate of the audio that is sent to the DAC is the same as the sampling rate coming out of the SPDIF output: 48khz. In other words, it goes from 48khz (ADC) -> higher sampling rate -> 48khz (DAC).
So just because these higher sampling rates are used for the processing stages doesn't mean it would be trivial to send a higher rate to the SPDIF output. The 48khz signal would need to be sample rate converted (SRC) at the output stage by a hardware SRC chip and Cliff's whole point is that software SRC's provide better quality than what is available with hardware SRC's.
The Axe-FX uses higher sampling rates (oversampling) during the processing stages. This is how it avoids aliasing when non-linearities are applied. But the sampling rate of the audio that is sent to the DAC is the same as the sampling rate coming out of the SPDIF output: 48khz. In other words, it goes from 48khz (ADC) -> higher sampling rate -> 48khz (DAC).
So just because these higher sampling rates are used for the processing stages doesn't mean it would be trivial to send a higher rate to the SPDIF output. The 48khz signal would need to be sample rate converted (SRC) at the output stage by a hardware SRC chip and Cliff's whole point is that software SRC's provide better quality than what is available with hardware SRC's.
I'm sorry but it doesn't explain the reason of the 48khz, this just confirms that Cliff has talked about oversampling somewhere.
If you do oversampling to avoid aliasing, you have at a time to do downsampling. Why to offer only 48khz? A less expensive DAC perhaps?
The Axe-FX uses higher sampling rates (oversampling) during the processing stages. This is how it avoids aliasing when non-linearities are applied. But the sampling rate of the audio that is sent to the DAC is the same as the sampling rate coming out of the SPDIF output: 48khz. In other words, it goes from 48khz (ADC) -> higher sampling rate -> 48khz (DAC).
I'm not buying that. Oversampling is usually applied at the AD/DA stage not during processing. Whatever bandwidth is gained when the source is sampled is filtered down to 24kHz (48kHz sample rate) at the input and not changed except maybe during the DA conversion. I'd bet the 24bit word length is increased to 32 or maybe 64 bits for processing. Unnecessarily increasing sample rate (especially for guitar signals) will cost big processing efficiency. I recall Cliff saying other processors employ SRC to 'offer' higher sample rate outputs but the processing is not really occurring at that rate. This SRC process is usually done on the cheap and most likely will add noise or artifacts anyway.
If you want to record your AxeFX at 96kHz, knock your self out!
I'm not buying that. Oversampling is usually applied at the AD/DA stage not during processing. Whatever bandwidth is gained when the source is sampled is filtered down to 24kHz (48kHz sample rate) at the input and not changed except maybe during the DA conversion. I'd bet the 24bit word length is increased to 32 or maybe 64 bits for processing. Unnecessarily increasing sample rate (especially for guitar signals) will cost big processing efficiency. I recall Cliff saying other processors employ SRC to 'offer' higher sample rate outputs but the processing is not really occurring at that rate. This SRC process is usually done on the cheap and most likely will add noise or artifacts anyway.
If you want to record your AxeFX at 96kHz, knock your self out!
so...are you saying record it in 48khz (default) then upsample it to what you're using on your recording AFTER you've recorded ...? this is hard to follow =0
Hi Cliff. I read some pages ago that you have changed how the usb will show up in the Daw. It seems to be now 4 in 2 out.
Is it possible at this point to reamp 2 different parts or guitars? I mean, since it is possible to have 2 amps at once, it is possible to have one that goes, let's say, input 1 and the other to input 2?
Thanks
1) Will there be tools available to capture/create IR's (or inbuilt into AxeEdit) ?
I know if I have the wav file then I can create II-Verion IR's with the tool from 1) - but sometimes I have only the I-Version Files:
2a) Can the old I-Version IR's be loaded into the II?
2b) Will there be tools available to convert I-Version IR's into II-Version ones ?
1) Will there be tools available to capture/create IR's (or inbuilt into AxeEdit) ?
I know if I have the wav file then I can create II-Verion IR's with the tool from 1) - but sometimes I have only the I-Version Files:
2a) Can the old I-Version IR's be loaded into the II?
2b) Will there be tools available to convert I-Version IR's into II-Version ones ?
When holding one of the "X-Y" buttons for a little longer (and not quickly releasing it) they should have a second function. Would it be possible to have them set-up as a forward-backward stepping between the effect blocks? (like the edit button is now a forward step button). I feel this would be a huge workflow enhancement when not using the editor. Hmm?
There are those of us who LOVE using loopers during our live setups. It was a little disappointing to hear the only differences on the Axe-Fx ll looper are the recording's length (60 secs) and that it will now have its own block.
Is there any way of having these features with the AXE-FX ll in a future FW release?
1) Quantization of recorded loops
2) Stereo loops
Memory-wise, would it be possible to put a common USB thumb drive to capture all recorded loops? This would seem like an easier solution than to add internal memory components.
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