Having worked with analog audio and equipment for many years, we have gained an expertise and comfort level that is admirable. We can hear noise and distortion and link these audible symptoms to likely causes, making the needed changes. We have our trusty tone generator and oscilloscope for isolating those more difficult challenges. We know how to measure levels, frequency response and distortion to gauge the audio performance.

Figure 1. Structure of AES digital audio. Click here to enlarge this image.

Unfortunately digital audio changes all this. The old cause and effect relationship we are familiar with is gone. Likewise, our trusty tone generator and oscilloscope offers little help in diagnosing a digital signal or equipment defect.

Transitioning from analog to digital audio analyzing and troubleshooting is not as hard as you may think. First, you need to study what is in the digital bits. Second, it is going to require some new and exciting analyzing tests and instruments so you can analyze those digital bits.

Digital audio bit by bit

An analog-to-digital converter changes the audio signal to digital values by sampling the audio level at fixed intervals of time. Sampling is like taking snapshots of the analog audio signal level over time. Sampling happens at equally separated intervals measured in the number of samples taken every second, expressed in hertz (Hz) or in thousands of hertz (kHz). Digital audio is commonly sampled at 44.1kHz or 48kHz or at doubled rates of 88.2kHz or 96kHz for professional recording.

Assigning a digital value to the audio level at each sample interval is called quantization. This requires that the amplitude range of the audio waveform be divided into level steps. A quantized binary value encoding system, Pulse Code Modulation (PCM), has been adopted for overall improved system performance. PCM quantifies linearly all quantizing intervals by means of a fixed scale over the signal amplitude range. PCM makes use of a two's complement system to distinguish positive and negative binary coded values. (See Figure 3.)

Figure 2. Clock/data/sample rate relationships in a 48kHz digital audio signal. A typical clock is 256 times the audio sample rate and the data bit-rate is 1/4 the clock frequency. Click here to enlarge this image.

The number of bits used to form the PCM digital words (bytes) that are used to represent each of the sampled audio levels can vary from eight to 24 bits. The bit word length determines the number of quantizing level steps (resolution) and the dynamic range. Each bit provides about 6dB of range. An eight-bit digital audio word length provides 48dB of dynamic range (quiet to loud audio range) while 16-bit provides 96dB and 24-bit provides 144dB.

In a digital audio system, the maximum audio level corresponds to 0dBFS (dB full scale), which is assigned the largest digital code word. Manufacturers have adopted the familiar zero VU level equal to +0dBu as a standard operating level (SOL). This level corresponds to -20dBFS, in which the digital values are well below the largest digital code word value. This provides 20dB of range for audio peaks to go above zero VU before digital clipping occurs.

PCM digital data is encoded using a second scheme called bi-phase mark coding (BPM). Bi-phase coding ensures a dc balanced data line, as each bit begins with a transition and ends with a transition. If the data bit is a “1,” a transition also occurs in the middle of the time slot. A data “0” has only the transitions at the beginning and end of the time slot and does not have a transition in the middle. Bi-phase coding doubles the data rate or frequency, as each data bit has two time intervals (clock cycles). A balanced line enables the receiver to properly detect logic high and low levels and the transition between them.