The Audio Menu is used to select and configure various audio I/O parameters and values.
Note - Some of the options displayed in the Audio Menu are contextual and will change depending upon the type of audio input selected. Therefore, the figures below may not exactly match what is available on the front panel of your unit.
Audio input options include Analog, AES67, Livewire, and AES3. The up/down arrows scroll through the options. In this and in subsequent edit screens, the Checkmark button accepts and applies any changes while the X button cancels without saving.
When using an AES67 audio source, the multicast address of the receive channel is set here. The up/down arrows change the number by a value of 1, and the left/right arrows navigate between fields.
When using a Livewire audio source, the receive channel is set here. The up/down arrows change the number by a value of 1, and the left/right arrows change the channel number by a value of 100.
When using an analog audio source, this value should be set to the absolute loudest level PDM II will see in your installation. The up/down arrows change the value. The "L" and "R" values represent the resulting digital levels for the incoming analog signal and can be used along with an externally generated test tone to adjust the sensitivity settings.
For most installations where 0 VU = +4 dBu, the recommended setting is +14 dBu = 0 dBFS.
Note: Don't confuse dBu with dBFS!
dBu - an analog measurement - is a comparison to an arbitrary voltage. When a console's VU meter reads 0, the output is typically +4 dBu, or 1.228 volts when using a sine wave. However, actual audio levels are typically much higher due to audio peaks in the waveform. Analog audio is very forgiving of such peaks, but digital audio is not.
A digital level of 0 dBFS (decibels in reference to full scale) is an absolute ceiling, and audio that exceeds that level will result in audible distortion. PDM II has a built-in limiter to prevent levels from exceeding 0 dBFS, but the recommended setting of 14 dBU = 0 dBFS provides 10dB of headroom while still maintaining an 84dB signal-to-noise ratio. Lower settings can be used when dealing with heavily processed input content, while higher settings may be necessary for content with very wide dynamic range.
The Analog Output Level screen displays how the analog output level reflects digital audio levels within PDM II's processor. With the setting as shown below, 0 dBFS within PDM II will yield a +14 dBu analog output. The output levels should generally be set to match the Input Sensitivity level, though levels can be adjusted to create a gain or loss at the analog outputs. Note that in Bypass mode, any such gain changes are also bypassed and the output levels will equal the input levels.
The default sample rate of the AES3 output can be set to 32, 44.1, or 48kHz.
The sample rate of the AES3 output can be locked to the sample rate of the AES3 input, even if the unit is not configured for an AES3 input. If there is no valid clock at the AES3 input, the output will revert to the rate set in the AES3 Default Output Rate screen above.
Audio over IP (AoIP) audio can be sent from PDM II as either a standard AES67 stream or as a Livewire stream. It can also be disabled if desired.
The multicast address of the AES67 transmit channel is set here. The up/down arrows change the number by a value of 1, and the left/right arrows navigate between fields.
The Livewire channel number for PDM II's audio output is set here. The up/down arrows change the number by a value of 1, and the left/right arrows change the channel number by a value of 100.
