Features
| Colour | Black |
|---|---|
| Brand | Matrix Audio |
Product Description
Matrix Audio S/PDIF 3 converter is designed to take a USB input from your PC, tablet or smartphone and convert it into a high quality digital signal. This signal can then be connected to a DAC, via an optical or coaxial lead.
The aim is to isolate the digital signal from the noise of the USB input. In fact, the ground wire of the USB input and the output is totally isolated.
The unit supports up to 32-bit/768kHz PCM input signals, while the output is a 1-Bit/DSD512 digital signal, of excellent quality.
High-accuracy femtosecond clock
The heart of the unit is the FPGA audio processor chip. This IC is driven by a very accurate femtosecond clock. Moreover, the IC performs frequency division from the 44.1kHz and 48kHZ base sampling frequencies.
As a result, decoding is very accurate. Additionally, the power supply is an LDO (Low DropOut) type. The clean power supply aids reproduction quality.
What is IIS-LVDS?
The X-SPDIF3 can be configured for four kinds of IIS-LVDS transmission standards. IIS is a circuit-level audio data interchange format. This is the purest form of digital audio data. LVDS stands for Low Voltage Differential Signalling. This is how the audio data is sent, and there are different formats. The X-SPDIF3 has a switch on the base that can match the DACs format requirements.
External power supply
The X-SPDIF3 is dual-powered, in that it can be powered from the connected device via USB. It can also be powered by an external power supply. This is needed if you are using a smartphone and you want to save the battery. An LED indicator indicates where the power is being sourced from. The USB power is cut off when an external power supply is connected.
Connectivity and build quality
The chassis of the Matrix Audio S/PDIF 3 Converter is made of a CNC-milled piece of aluminium. This provides good electromagnetic shielding as well as providing vibration resistance. The case has practicality as well as good looks and will isolate any interference from peripheral devices. There are two inputs, one for the USB audio input and one for the external power supply. Additionally, there are three digital outputs, a coaxial and optical as well as a S/PDIF connector.
What is DSD?
With CDs, the audio is encoded using two methods. The first is the amplitude. This is sampled using 16-bits giving 65,536 possible levels. The other is the sampling frequency, or how often each 16-bit ‘sample’ is taken. In the case of a CD it is 44.1kHz. This was chosen so that the sampling frequency is at least double the highest audio frequency of 20 kHz. The CD encoding method is known as PCM or Pulse Code Modulation.
DSD or Direct Stream Digital uses a different approach. It was originally used to archive old analogue recordings. DSD uses a simpler and more space-efficient way of encoding digital music, than PCM. However, the sampling rate is still based on multiples of 44.1kHZ. Notably, DSD is also used for SACDs.
Interestingly, DSD uses only one data bit to encode the music. This bit indicates whether the next sample is higher or lower than the previous one. For example, if it is higher a 1 is output. This is achieved by sampling the waveform at a very high frequency, in fact it is 64 times the sampling rate of a CD. Hence the need for a femtosecond clock in the X-SPDIF3. The advantage of DSD is its simplicity.
