Digital audio: USB, SPDIF, AES
From Axe-Fx II Wiki
The information on this page supplements the official manuals.
Digital audio: supported by which Fractal Audio products?
- Axe-Fx II: yes (USB Audio, SPDIF, AES).
- AX8: yes (SPDIF output only).
- FX8: no.
Why use digital audio?
A digital connection skips the analog/digital conversion. The high-quality A/D converters in the Axe-Fx II and AX8 for the analog outputs also deliver high-quality results.
Sample rate: fixed at 48kHz
The sampling rate of the Axe-Fx II, FX8 and AX8 is fixed at 48kHz (24-bit). You need to set digitally connected devices to the same sampling rate. Resampling is better handled by software.
Make sure to set the sampling rate in your DAW software and system settings to 48kHz. Example
- "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."
- "IMHO, the ideal sample rate is 64 kHz but that's not a standard. The nice thing about 64 kHz is that you can have a gentle transition band from 20 kHz to Nyquist which results in shorter filters, lower latency, less phase shift, etc. I was very tempted to make the Axe-Fx II run at 64 kHz but people probably would have freaked out." source
- "48 kHz is considered "pro" sampling rate. The reason for 44.1 kHz on CD's is subject to debate. Some maintain that the sample rate was lowered so that Beethoven's 9th would fit on a single CD. Others claim that it was because that rate was compatible with video equipment. IMO 44.1 kHz is insufficient for professional audio. Personally I would prefer 64 kHz. Whilst Nyquist theorem is all well and good most people don't understand the details and simply state "the sample rate must be twice the highest desired frequency". The problem with this is as you approach Nyquist the filter demands become extreme. The more extreme the filter demands the more taps are needed, the more precision is needed, the more latency is incurred, etc. A 64 kHz sample rate would give you a nice, smooth roll-off from 20 kHz to 32 kHz rather than the brick wall you get with 44.1 kHz. There is no hardware advantage to using 48 vs. 44.1. The costs would be the same in either case. Modern converters use over-sampling techniques to implement the necessary anti-aliasing filters thereby reducing off-chip filtering to simple circuits. MP3s have no native sample rate but are typically 44.1 kHz because they are usually derived from CDs. MP3 is a psycho-acoustic compression format that exploits frequency masking to lower the data required to store audio information." source
Digital audio and latency
When monitoring the audio signal through the computer's output, the latency depends on the computer. When monitoring audio via the hardware, latency is none.
Axe-Fx II and USB
Read this: USB.
The Axe-Fx II is an Audio Class 2.0 compliant device. A class-compliant device requires no drivers. The drivers are provided by the OS manufacturer. Audio Class 2.0 also encompasses MIDI-over-USB. Microsoft does not support Audio Class 2.0. Therefore FAS provides a driver for Windows systems. The driver for Windows contains both the firmware installer and the audio drivers.
Apple does support Audio Class 2.0, but poorly. To overcome this you can increase the buffer size in the Axe-Fx II's I/O > Audio menu. The driver for Macs is NOT an audio driver. It is a firmware installer. The Axe-Fx II uses a "soft" USB controller. It gets its code from the host computer. When you turn the Axe-Fx II on it requests firmware from the host. This is superior to a hard-coded controller in that updates merely require a new host image rather than reflashing the controller.
Use the USB buffer parameter in I/O > Audio to lower values for less latency, set to higher values if experiencing distorted audio. Low values generally work fine with Windows machines. OS-X computers usually need higher values due to poor clock adaptation. You should stop USB audio streaming when changing this value so as to allow the buffer to reset properly. Streaming can be stopped by closing the application sending data to the Axe-Fx or by disconnecting the USB cable.
The “USB” bar graph in Utility > Status displays the amount of data in the USB FIFO buffer. Ideally the bar should be at around 50%. If the bar sinks all the way to the bottom or goes all the way to the top, then the buffer may under/overflow and the USB buffer size should be increased. The number of buffer errors that have occurred since the last buffer reset is indicated above the bar graph.
USB Level in I/O sets the level of the USB input signal sent to the main outputs. If you don't hear anything when monitoring the Axe-Fx II through a computer, check this parameter. Also verify the USB/DIGI OUT setting.
Linux: read this.
Axe-Fx II and USB Audio
USB Audio rate is fixed at 48 kHz.
- "The Axe-Fx II USB is 24 bits. This is 144.7 dB of dynamic range. Full-scale is about +20 dBu. So even if your guitar is -20 dBu (-40 dB re. FS) you still have over 100 dB of dynamic range. A typical single coil pickup can easily exceed -20 dBu. A humbucker can easily exceed 0 dBu. Full-scale of 20 dBu gives you a few bits of headroom in case of very hot pickups. The self noise of a guitar pickup and associated electronics limits its dynamic range to less than 100 dB typically." And: "The digital bit depth on the USB and Digital I/O exceeds both the dynamic range of the Axe-Fx itself and certainly that of any guitar. Furthermore the bit depth is sufficient to fully capture the dynamic range of a guitar while still maintaining +20 dBu as full-scale." source
- "The hardware is incapable of doing 4x4. The only choices are 3x3 or 4x2 and Logic doesn't work with 3x3. We also had some issues with 3x3 in Windows 7 IIRC." source
Simultaneous use of analog and digital audio connections
You can use digital input together with analog Output 1 (and simultaneous digital out). And you can use Input/Output 2 while using the digital input. source
Disable the analog output signal when using digital audio
When the Axe-Fx II is connected to a DAW through USB, you're using monitors connected to the Axe-Fx and you're recording, you may want to monitor just the DAW signal, not the signal from the Axe-Fx. To accomplish this, turn down the preset level. Or set USB/Digi Out Source to Out 2, then route the guitar to FXL only. source
Axe-Fx II as master or slave
- "The Axe-Fx II can be a slave. Set the Input Source to AES. It will derive its internal clock from the input stream. The input stream must be 48K. Note that SOMETHING must be the master in this case." source1 source
- "The Axe-Fx will derive its clock from the AES/SPDIF when using Digital In. In Analog In it uses its internal clock." source
Word Clock (Axe-Fx II only) selects the clock source for the A/D and D/A converters as follows:
- Auto: uses the internal clock if the input source is Analog or USB. Uses the recovered SPDIF/AES clock if the input is SPDIF/AES.
- SPDIF/AES IN: uses the recovered clock for all input sources. A valid 48 kHz data stream must be present at the AES or SPDIF input. If a valid stream is not detected the unit will fall back to the internal clock and display "NO INPUT CLOCK!". The SPDIF/AES select must be set to the appropriate value, i.e. if the data stream is input to the XLR jack then SPDIF/AES SELECT must be set to AES.
- Fractal Audio:
- "Set Word Clock to SPDIF/AES In. Connect a cable from the ULN-8 to AES In or SPDIF In. Set SPDIF/AES Select to appropriate input used." source
Axe-Fx II: configure your Axe-Fx II for S/PDIF.
AX8: provides SPDIF output only. The strength of the SPDIF signal level depends on the position of the front panel output knob (unlike the Axe-Fx II). Fractal Audio: "The SPDIF is a digital representation of OUTPUT 1." source
Use the Axe-Fx II or AX8 as an analog-to-digital converter
To use the device as an analog-to-digital converter (source), do this:
- Create a preset with nothing but shunts from input to output.
- Or set Input 1 Left Select to to Rear and plug the device into Input 1 Left on the back (Axe-Fx II).
- Or connect In and Out and engage Bypass Mode.
Connecting the Axe-Fx II to a DAW drops the volume or puts the Axe-Fx II into Bypass mode
Some DAWs send MIDI commands (intended for other devices) which lower the volume of the Axe-Fx II or put it into Bypass mode. To solve this, adjust the DAW settings. If not possible, force the Axe-Fx II to ignore those commands by setting MIDI CCs 10 and 13 in I/O > Control to "none".