AD1953_15 datasheet(15/36 Pages) AD

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REV. 0

AD1953

–15–

The spread_level is a linear number in 2.20 format that multiplies

the processed left-right signal before it is added to or subtracted

from the main channels. The parameter alpha_spread is related to

the cutoff frequency of the first-order low-pass filter by the equation

Alpha spread

EXP

spread

freq

_. –

–.

××









where EXP is the exponential operator, spread_freq is the low-pass

cutoff in Hz, and fS is the audio sampling rate.

Note that the stereo spreading algorithm assumes that frequencies

below 1 kHz are present in the main satellite speakers. In some

systems, the crossover frequency between the satellite and

subwoofer speakers is quite high (> 500 Hz). In this case, the

stereo spreading algorithm will not be effective, as the frequencies

that contribute to the spreading effect will be coming mostly from

the subwoofer, which is a mono source.

Delay

Each of the three DAC channels has a delay block that allows

the user to introduce a delay of up to 165 audio samples. The

delay values are programmed by entering the delay (in samples)

into the appropriate location of the parameter RAM. With a

44.1 kHz sample rate, a delay of 165 samples corresponds to a

time delay of 3.74 ms. Since sound travels at approximately

1 foot/ms, this can be used to compensate for speaker place-

ments that are off by as much as 3.74 feet.

An additional 100 samples of delay are used in the look-ahead

portion of the compressor/limiter, but only for the main two

channels. This can be used to increase the total delay for the left

and right channels to 265 samples, or 6 ms at 44.1 kHz.

Main Compressor/Limiter

The compressor used in the AD1953 is quite sophisticated and

is comparable in many ways to professional compressor/limiters

used in the professional audio and broadcast fields. It uses rms/

peak detection with adjustable attack/hold/release, look-ahead

compression, and table-based entry of the input/output curve for

complete flexibility.

The AD1953 uses two compressor/limiters, one in the subwoofer

DAC and one in the main left/right DAC. It is well known that

having independent compressors operating over different frequency

ranges results in a superior perceived sound. With a single-band

compressor, loud bass information will modulate the gain of the

entire audio signal, resulting in suboptimal maximum perceived

loudness as well as gain pumping or modulation effects. With

independent compressors operating separately on the low and

high frequencies, this problem is dramatically reduced. If the

AD1953 is being operated in 2-channel mode, an extra path is

added so that the subwoofer channel can be added back into the

main channel. This maintains the advantage of using a 2-band

compressor, even in a 2.0 system configuration.

Figure 7 shows the traditional basic analog compressor/limiter.

It uses a voltage controlled amplifier to adjust gain and a feed-

forward detector path using an rms detector with adjustable

time constants, followed by a nonlinear circuit to implement the

desired input/output relationship. A simple compressor will have

a single threshold above which the gain is reduced. The amount

of compression above the threshold is called the compression

ratio and is defined as dB change in input/dB change in output.

For example, if the input to a 2:1 compressor is increased by

2 dB, the output will rise by 1 dB for signals above the threshold.

A single “hard” threshold results in more audible behavior than

a so-called “soft-knee” compressor, where the compression is

introduced more gradually. In an analog compressor, the soft-knee

characteristic is usually made by using diodes in their exponential

turn-on region.

FILTER

RMS DETECTOR

WITH dB OUT

COMPRESSION

CURVE NON-

LINEAR CIRCUITS

THRESHOLD

SLOPE

VCA WITH EXP

CONTROL

OUT

Figure 7. Analog Compressor

The best analog compressors use rms detection as the signal

amplitude detector. RMS detectors are the only class of detec-

tors that are not sensitive to the phase of the harmonics in a

complex signal. The ear also bases its loudness judgment on the

overall signal power. Using an rms detector therefore results in

the best audible performance. Compressors that are based on

peak detection, while good for preventing clipping, are generally

quite poor when it comes to audible performance.

RMS detectors have a certain time constant that determines

how rapidly they can respond to transient signals. There is always

a trade-off between speed of response and distortion. Figure 8

shows this trade-off.

INPUT WAVEFORM

COMPRESSOR ENVELOPE –

FAST TIME CONSTANT

COMPRESSOR ENVELOPE –

SLOW TIME CONSTANT

Figure 8. Effect of RMS Time Constant on Distortion

In the case of a fast-responding rms detector, the detector enve-

lope will have a signal component in addition to the desired dc

component. This signal component (which, for an rms detector,

is at twice the input frequency) will result in harmonic distortion

when multiplied by this detector signal.

The AD1953 uses a modified rms algorithm to improve the

relationship between acquisition time and distortion. It uses a

peak-riding circuit together with a hold circuit to modify the rms

signal, as shown in Figure 9. Figure 8 shows two envelopes—one

with the harmonic distortion and another, flatter envelope,

which is produced by the AD1953.

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