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5. Sampling issues

Until the late 19th century, audio systems for live and recording applications were designed using acoustical and
mechanical tools - such as the walls of a concert hall (amplification, colouring), copper tubes in shipping (short
distance transport), grooves on a wax roll picked up by a needle and amplified by a big horn (long distance transport).
In 1872 everything changed when, independently from each other, Emil Berliner and Thomas Edison invented
the carbon microphone(*5A), introducing the possibility of transforming acoustic waves to electronic signals.
This was the beginning of the ‘analogue’ era - using electrical circuits to mix, amplify, modify, store and transport
audio signals. This era saw the introduction of the electrical Gramophone in 1925 (Victor Orthophonic Victrola*5B),
the reel to reel tape recorder in 1935 (AEG’s Magnetophon*5C) and the compact cassette in 1962 (Philips*5D). In
the years to follow, large scale live audio systems became available built around mixing consoles from Midas,
Soundcraft, Yamaha and others.

5.1 Digital Audio

In 1938, Alec Reeves from ITT patented Pulse Code Modulation (PCM)(*5E), marking the start of the development of digital systems to process and transport audio signals - with the goal to make systems less susceptible for noise and distortion. In 1982, the first album on Compact Disc, developed by Philips and Sony, was released (Billy Joel’s 52nd street) - introducing 16-bit 44.1 kHz PCM digital audio to a broad public(*5F).

The first mass produced Yamaha digital mixing system was the DMP7, launched in 1987 - introducing 16-bit A/D and D/A conversion and Digital Signal Processing (DSP) architecture to small scale music recording and line mixing applications(*5G). But where 16 bits are (only just) enough for CD quality playback, the 90dB typical dynamic range it can reproduce is insufficient for large scale live mixing systems. In 1995, the Yamaha 02R digital mixing console introduced 20-bit A/D and D/A conversion, supporting 105 dB dynamic range(*5H). Matching - or in many cases surpassing - the audio quality of analogue systems at that time, the 02R was the trigger for both the recording and the live audio market to start a massive migration from analogue to digital. Since then, A/D and D/A conversion technologies have matured to 24-bit A/D and D/A conversion and transport, and 32-bit or more DSP architecture, supporting system dynamic ranges of up to 110 dB - far beyond the capabilities of analogue systems(*5I). DSP power has reached levels that support massive functionality that would have been simply not possible with analogue systems.

At the time of writing of this white paper, the majority of investments in professional live audio systems involve a digital mixer. Also, networked distribution systems to connect inputs and outputs to mixing consoles are now replacing digital and analogue multicore connection systems.

All networked audio systems use A/D converters to transform analogue audio signals to the digital domain through the sampling process, and D/A converters to transform samples back to the analogue domain. Assuming that all A/D and D/A converters, and the distribution of the samples through the digital audio system, are linear, sampling does not affect the Response of a digital audio system - all A/D and D/A processes and distribution methods sound the same. (only intended changes - eg. sound algorithm software installed on a system’s DSP’s - affect the sound). Instead, sampling and distribution can be seen as a potential limitation as it affects audio signals within the three audio dimensions: level, frequency and time. This chapter presents the sampling process limitations to a system’s Performance: dynamic range, frequency range and temporal resolution.

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