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5. Sampling issues

5.8 Clock phase

All processes in a digital audio system are triggered by a common synchronisation signal: the master word clock. The rising edge of the word clock signal triggers all processes in the system to happen simultaneously: A/D and D/A conversion, DSP processing and the transmission and receiving of audio data (transport).

For packet switching audio protocols using star topology networks (such as Cobranet, Dante, AVB), the distribution of the synchronisation information uses separately transmitted timing packets, ensuring that all devices in the network receive their clock within a fraction of the sample time (20.8 microseconds at 48 kHz). This assures for example that all AD samples in the system end up in the same sample slot. But for streaming audio protocols using daisy chain or ring topologies it is different: the clock information is represented by the audio data packets, so the length of the daisy chain or ring determines the clock phase caused by the network. For example, Ether- Sound adds 1,4 microseconds latency per node - accumulating to higher values when many nodes are used.

Additionally, a device that synchronises to an incoming synchronisation signal needs some time to do so. Digital circuit designers focus primarily on matching the incoming synchronisation signal’s frequency as stable, flexible and fast as possible, rather than on the clock phase. As a result, digital audio devices on the professional audio market all have a different clock phase, which is almost never documented in the product specifications. Figure 514 presents a system that uses two different A/D converters that have a different clock phase: the samples are taken and sent to a digital mixer at a different time. As soon as the samples arrive in a digital mixer, the receiver circuit in the mixer will buffer and align the samples, mixing them as if they were taken at the same time, introducing a latency difference between the samples that is smaller than one sample. Because the latency difference caused by differences in clock phase are smaller than one sample, digital delays can not be used to compensate them - the minimum delay time available in a digital signal processor or mixer is one sample.

The compensation of clock phase differences can not be included in the system engineering phase of a project because almost all digital audio devices on the market do not specify clock phase delay. Conveniently, in practise, acoustically tuning a system (placing the microphones and speakers) already includes the compensation of all acoustical delays as well as clock phase; aligning all signal paths to produce a satisfactory audio quality and sound quality according to the system technician. As it is important not to move microphones and loudspeakers after a system has been acoustically tuned, it is of equal importance not to change routing and word clock distribution. Some ‘internal/external clock’ comparison listening sessions change the word clock distribution topology manually. The resulting differences in sound are often attributed to jitter performance, while in some cases clock phase changes might have been more significant: a clock phase difference close to half a sample can already be detected both in timing and comb filter effect (see also figure 511 in chapter 5.6).

>>5.9 Temporal resolution

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