To switch the DAC on press the “ON/OFF” button on the front panel. The red standby
indicator blinks green for approx 10 sec. whilst the unit warms up. When ready, the
indicator will stop blinking, turn green, and the last active source will be selected
automatically. The front panel display will be on while the unit is executing a command
and then turns off after a short timeout period. When an input signal is detected, the
frequency and bit depth will be read out on the display.
Re-clocking
– Enable this function in the user menu (see “Menu structure” below) . This very
important feature of the DAC allows for all jitter to be removed from the input source.
Data is read onto the device’s memory and then independently read out using a ultra
stable clock. When enabled, this option will completely replace the incoming clock with
an ultra low jitter TCXO based clock. The DSP monitors the incoming sample frequency
and detects standard sample rate signals - 44.1 kHz, 48 kHz, 88.2 kHz, 96 kHz, 176.4 kHz, 192
kHz, 352.8 kHz and 384 kHz. The on-board clock then completely replaces the incoming
clock. The source’s clock is used for other sampling frequencies. The DSP allocates a
huge internal FIFO buffer (1/2 second at 44.1), that stores the incoming audio to
decouple the incoming and outgoing data streams. Long absolute digital silences in the
music stream, such as between tracks and during pauses, are selectively shortened or
lengthened by the DSP to maintain data synchronization. This results in a significant delay
between the audio source and the analog audio. You will not normally notice this delay
unless video is synchronized with the audio. For this reason you may want this feature to
be turned off when watching video, or the video might be delayed.
Up-sampling
– Enable this function in the user menu (see “Menu structure” below) thus all input signals
will be up-sampled to max resolution 32/352 or 32/384 depending on input signal. To
achieve this, a digital filter takes a look at a window of the music being played, and
because that music was received from a digital source, there are holes in it, between
data bytes. The filter looks at the shape of the signal in the window and tries to figure out
what the missing data is. The bigger the window, the better a job it does, and the bigger
processor is needed. You can select which up-sampling algorithm to use (UPS F1 or F2 in
the User Menu) in the process of filling in the gaps between the data.
Digital Filter
– The digital filter is necessary because mirrored image frequencies created during the
conversion process must be removed. If the DAC did not have a digital filter, an analog
filter with an aggressive response must remove these image frequencies. Analog filters
seriously damage the signal by corrupting the original phase of the sound and cannot
fully remove the high frequency images. This results in harsh or rolled off high frequencies
and poor soundstage focus.
OPERATION
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