The design work on the Alpha-X HDM01 is progressing well. This past weekend I did the necessary to get the digital audio path up and running. Even though many books and other educational resources tend to steer clear of this subject, its really not that difficult or mysterious.
The screenshot below is what it looks like when audio is played back, with a sample rate of 48kHz at 24 bits. Indeed even my humble oscilloscope (25MHz) is fast enough to capture this in real time. The upper trace is LRCK, and the lower trace is the PCM data, all 24 bits of it.
The above was captured during playback of this album (track 1):
A note for the audiophiles reading this blog post:
The LRCK is measured by the 'scope as 47.98kHz. This is NOT an accurate measurement, the Rigol oscilloscope never excelled at frequency measurement hence why we use a dedicated frequency counter. So there is no argument here for "jitter". At any rate, this system uses oversampling, we cannot reasonably expect to install non-oversampling D/A converters in this system, because a) there is no space, and b) that would raise the cost of this system into the stratosphere, and c) such devices are not really offered anymore due to cost and low demand (Surprise- semiconductor companies are businesses too, complete with suits and corporate greed!)
A note about jitter in digital audio:
In digital audio, jitter refers to the deviation of the clock source from the ideal, or nominal rate. In the early days of digital audio this was a fuck-up! Jitter meant the D/A converter began to do "stuff" it wasn't supposed to, leading to quantization noise and other "artifacts" that one might have experience with when cueing or mucking about with heavily compressed MPEG sources.
In systems such as these, especially in 2017, the best D/A converters in the world are of the Sigma-Delta modulated type, which is, by virtue of its inherent design, largely unaffected by jitter. The sampling rate is insanely high, rendering any harmonics and intermodulation products virtually non-existent in the audio band. Indeed, the low-pass filter required for the D/A converter chip is a simple pi-filter. No need to go the whole hog with Butterworth or Chebyshev, raising the cost because of the need of expensive WIMA capacitors and E96 series resistors.
Also, we have now got the benefit of some 30+ years of DIR (digital interface receiver) design behind us, meaning we have clever electronics that can tolerate an insane amount of distortion in the digital audio path and automagically a beautifully stable and jitter-free clock is obtained.
It is my belief that many audiophiles are wasting a lot of their money pursuing snake oil with fancy gold plated S/PDIF cables and other contrived items. In this exercise I have managed to obtain the sound I wanted, that often-elusive studio quality, on a budget, and on a fucking breadboard! The difference is that I know what I am doing!