DAC

Digital audio is PCM (Pulse Code Modulation) audio most of the time.
It consist of two components, the value of the signal (represented by 16 or 24 bits words) and the time step (sample rate).
We have two components, the signal and the time.

Sounds logical but pretty often you hear the 'bits are bits' theory, implying that if the bits are right, everything is right. This theory leaves the other half, the time step, out of the equation.

To play PCM audio, the bits has to be translate to a equivalent voltage and this must be done with a time step matching the sample rate.

This is done by a Digital to Analogue Converter, a DAC for short.

 

The sample rate is generated by a clock.

As absolute perfection don't exist, there is always some fluctuation in clock speed.

This is called clock jitter.

Interface jitter is jitter introduced in the transmission of digital signals.

Jitter could be induced by noisy power supplies, improper ground path and electromagnetic interference.

Crucial is the sampling jitter, deviations in the sampling interval in the DA conversion stage

According to the Redbook audio standard the clocks frequency should be within +/-100ppm (parts per million).

A deviation of 100 ppm means that a 440Hz tone deviates +/- 0.044Hz.

NOS

The first generation was Non OverSampling.
The DAC simply runs at the speed of the incoming stream.
Inherent to the conversion is that higher-frequency multiples of the audible range are created, the so called aliases. In case of CD audio, the sampling rate is 44.1 kHz, the audible range is the half, so the first alias will start at 22.050 kHz.
To avoid that these aliases burn you tweeters, a low pass filter is required.
This filter starts at 20.000 and has to remove everything before the first alias starts (22.050 assuming CD audio) so it has to be very steep (brick wall filter). Filters this steep are expensive, complex and introduces all kind of artifacts like phase distortion and pre-ringing.

 

Schema of a NOS DAC

 

Most DAC’s today use oversampling to avoid the filter problem.

 

Schema of a modern DAC

 

• The receivers extracts the data (the signal) and the clock from incoming signal.

  In general this is S/PDIF over coax or Toslink. Modern DAC's offers USB input to.

• A digital filter might be used to do SRC (sample rate conversion)
• Some DAC's do reclocking, it makes them less sensitive to input jitter.
• The DAC chip converts the digits to a current
• The current is converted into a voltage (IV converter)
• This signal is filtered to get rid of the switching noise.
• The amplifier is needed to bring the signal up to line level (just like a phono stage).

There are other designs possible, an overview can be found here: http://en.wikipedia.org/wiki/Digital-to-analog_converter

€ 23.50	€ 19.500

A DAC might be a sound card or a separate box.

Building a DAC

Very few companies build there own DAC (the chip set)
dCS and Chord are the one’s I know using a FPGA to build their own converter.
Most companies use a chipset by Burr Brown (TI today), AKM, Wolfson, Analog Devices, ESS, etc.

Feeding the DAC

Ask in an audio forum “what is the best way to connect the PC to the audio” and the replies will probably be

• Take a USB DAC, there are some pretty good ones out there
• No, go FireWire, much better, USB sucks
• No, a good internal sound card equals  all these expensive external boxes at a fraction of the cost.
• Get a media center with SPDIF out.
• Get a Mac mini
• use a Squeezebox
• Etc.

Onboard sound card uses the PCI-bus. The analogue out in general is RCA, the pro-models have balanced out (XLR).

 

An outboard DAC can be connected using Toslink, SPDIF, AES/EBU, USB, FireWire, Ethernet, WiFi and then I don’t mention protocols like MADI, ADAT, CobraNet and other pro-stuff.

 

Toslink, SPDIF, AES/EBU are popular in the audio world.
It are unidirectional protocols, there is no 2-way communication.
The sender starts to stream in real time and the receiver has to lock on the incoming stream.
In principle, any variation in clock speed by the sender will result in input jitter at the receiver (the DAC). There are tricks to improve on this like ASRC.

 

USB en FireWire are typically computer protocols.
Both can be used in different modes.
In asynchronous mode the DAC times the bus.
In this scenario the clock of the DAC can run at a fixed speed as there is no need to sync on an incoming stream. This mode is the solution to eliminate input jitter.

 

Networking over Ethernet (wire) or WiFi (wireless) are asynchronous by design.
Packages of data are send using a protocol like TCP/IP.
This protocol is very strict, it is build with bit perfect transmission in mind.
It is a bi-directional protocol, if a package fails the checksum test, the receiver simply tells the sender so and it will get a new one.
All input is buffered.
Sound ideal, asynchronous, bit perfect transmission in other words a jitter free bit perfect connection. But you need a computer to do this and this raises a question: how to get the audio out of it.