AN1836-AN21161 Analog Devices, AN1836-AN21161 Datasheet - Page 5

• Manufacturer Part Number: AN1836-AN21161
• Description: Interfacing the ADSP-21161 SIMD SHARC DSP to the AD1836 (24-bit/96 kHz) Multichannel Codec
• Manufacturer: Analog Devices

0.2 The Benefits of 32-bit Audio Processing of 24-bit "professional-quality" audio
Today 16-bit, 44.1 kHz PCM digital audio continues to be the standard for high quality audio in most current applications, such
as CD, DAT and PC audio. When the compact disc was introduced in the early 1980s, audio designers elected to work with 16-
bit words sampled at 44.1 kHz for a mixture of technical and commercial reasons. Factors that limited their options included
the quality of available A/Ds, the quality and cost of other digital components, and the density at which available media could
store digital data. They also thought that the format would be sufficient to record audio signals with all the fidelity required for
the full range of human hearing.
But recent technological developments and improved knowledge of human hearing have created a demand for greater word
lengths in the professional audio sector. Research within the last decade indicates that the sensitivity of the human ear is such
that the dynamic range between the quietest sound the average person can detect and the maximum sound that person can
experience without pain is approximately 120 dB. Concerning word width, it’s clear that 16-bit CD-quality audio no longer
corresponds to the highest-quality audio a system should be able to store and play back. Digital converter technology has
advanced to where audio engineers can make recordings with a dynamic range of 120 dB or greater, but a compact disc is
unable to accurately carry them because the CD standard limits word size to 16 bits. Many manufacturers of pro equipment
have already developed new products using 24-bit conversion and 96-kHz sample rates. Many recording studios now routinely
master their recordings using 20-bit recorders, and are quickly moving to 24 bits. These technological developments are now
making their way into the consumer and so-called “prosumer” audio markets. The most evident consumer incarnation is DVD
which is capable of carrying audio with up to 24-bit resolution. New DVD standards are extending the digital formats to 24-
bits at sample rates of 96 kHz and 192 kHz formats. Other products include DAT recorders which can sample at 96kHz. Many
professional audio studio manufacturers now offer DAT recorders with 24-bit conversion, 96 kHz sampling rate.
In fact, three trends can be identified which have influenced the current generation of digital audio formats which are set to
replace CD digital audio, and these may be summarized as follows:
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The Analog Devices AD1836 offers 24-bit, 96 kHz multichannel audio capability to meet many new requirements in the
professional, consumer and automotive audio markets. Multibit sigma-delta converters such as the AD1836 are capable of 24-
bit resolution, capable of exceeding the 80 to 96 dB dynamic range available using 16 bit conversion. The popularity of 24-bit
D/As is increasing for both professional and high-end consumer applications. The reason for using these higher precision
converters for audio processing is clear: their distortion performance (linearity) is far superior than possible with 16-bit
converters. The other obvious reason is the increase in SNR and dynamic range.
Now consider the DSP word size with respect to the converter's word size for processing digitized audio samples. Figure 4
shows the capable dynamic ranges for DSPs with native data word widths of 16, 24, or 32-bit data (assuming fractional fixed-
point data types). Assuming 6 dB per bit, a 16-bit DSP can support dynamic ranges of 96 dB, a 24-bit can support 144 dB,
while a 32-bit DSP that has native fixed point support (such as the ADSP-21161) can support signals up to 196 dB.
In general, if a digital system produces processing artifacts which are above the noise floor of the input signal, then these
artifacts will be audible under certain circumstances, such as when an input signal is of low intensity or limited frequency
content. Therefore, the digital processing algorithm operating on the input A/D samples should be designed to prevent
processing noise from reaching levels at which it may appear above the converter's noise floor of the input and hence become
audible [3]. For a digital filter routine to operate transparently, the resolution of the processing system must be considerably
greater than that of the input signal so that any errors introduced by the arithmetic computations are smaller than the precision
of the ADCs or DACs. In order for the DSP to maintain the SNR established by the A/D converters, all intermediate DSP
calculations require the use of higher precision processing as recommended by Wilson, Dattorro, and Zolzer [3, 4, 5]. Some
manifestations of the causes of finite word length effects that can degrade an audio signal’s SNR are: A/D conversion noise,
quantization error of arithmetic computations from truncation and rounding, computational overflow and coefficient
quantization.
Higher resolution - 20 or 24 bits per word
Higher sampling frequency - typically 96 kHz
More audio channels