All input data, regardless of its format, is converted to 24 bits and asynchronously resampled at a very high rate. DSD is unpacked, converted to PCM and resampled. The resampled data enters a Digital Filter. The Digital Filter is configured in optimal mode for the output format of the Sample Rate Converter, therefore there are no user-selectable filter modes. Data then enters the DAC stage where it is converted into an analog signal.
Jitter essentially loses its meaning in this configuration. Input data jitter is practically irrelevant. The Asynchronous Sample Rate Converter takes care of proper word alignment and timing during conversion. The three digital blocks are synchronized to a single, high precision clock to ensure perfect timing. Any CD transport can be used with the DAC1, not requiring external re-clocking or clocking from the DAC1.
The actual DAC stage is a differential output (balanced) design. Its residual distortion is approximately 0.00035% at full scale output, consistent for all input formats. At typical output levels of -10dBFS the distortion falls off to 0.0001% (-120dB) and becomes essentially zero at levels below -20dBFS, “lost in the grass” as we say, at -130dB.
This matters! It’s during the quiet passages that DAC distortion becomes most noticeable. One bit is lost with every 6dB reduction in level, therefore a 16-bit DAC will be working effectively at 14-bit resolution or less most of the time. This is the reason why early designs created a bad reputation for digital audio. We hope to rectify this misunderstanding.
So, is 130dB enough? It is more than enough, and here’s why: a signal which is -130dB lower relative to the full scale output voltage of 2Vrms has an amplitude of 3 million times less, or 0.7 microvolts. That is less than many amplifiers’ input-referred noise voltage. We can confidently say that the DAC1 will have an insignificant noise contribution, if any, to the signal chain.
