AD1953YSTZRL Datasheet AD1953YSTZRL, AD1953YSTZ, AD1953YSTZRL7 | P16-P18

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

_.–

–. _

××

⎛

⎝

⎜

⎞

⎠

⎟

20 π

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

cutoff in Hz, and f

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.

REV. A

AD1953

–16–

INPUT WAVEFORM

HOLD TIME, SPI-

PROGRAMMABLE

RELEASE TIME, SPI-

PROGRAMMABLE

Figure 9. Using the Hold and Release Time Feature

Using this idea of a modified rms algorithm, the true rms value

is still obtained for all but the lowest frequency signals, while the

distortion due to rms ripple is reduced. It also allows the user to

set the hold and release times of the compressor independently.

The detector path of the AD1953 is shown in Figure 10. The rms

detector is controlled by three parameters stored in parameter

RAM: the rms time constant, the hold time, and the release rate.

The log output of the rms detector is applied to a look-up table

with interpolation. The higher bits of the rms output form an

offset into this table, and the lower bits are used to interpolate

between the table entries to form a high precision gain word.

The look-up table resides in the parameter RAM and is loaded

by the user to give the desired curve. The look-up table contains

33 data locations, and the LSB of the address into the look-up

table corresponds to a 3 dB change in the amplitude of the detec-

tor signal. This gives the user the ability to program an input/

output curve over a 99 dB range. For the main compressor, the

table resides in locations 110 to 142 in the SPI parameter RAM.

LOOK-UP TABLE

LINEAR

INTERPOLATION

MODIFIED RMS

DETECTOR WITH

LOG OUTPUT

OUTPUT TO

GAIN STAGE

HIGH BITS (1LSB = 3dB)

LOW BITS

TIME

CONSTANT

HOLD

RELEASE

Figure 10. Gain Derived from Interpolated Look-Up Table

One subtlety of the table look-up involves the difference between

the rms value of a sine wave and that of a square wave. If a

full-scale square wave is applied to the AD1953, the rms value of

this signal will be 3 dB higher than the rms value of a 0 dBFS sine

wave. Therefore, the table will range from +9 dB (location 142)

to –87 dB (location 110).

The entries in the table are linear gain words in 2.20 format.

Figure 11 shows an example of the table entries for a simple

above-threshold compressor.

INPUT LEVEL – 3dB/TABLE ENTRY

OUTPUT LEVEL – dB

INPUT LEVEL – 3dB/TABLE ENTRY

LINEAR GAIN

1.0

DESIRED

COMPRESSION

CURVE

Figure 11. Example of Table Entry for a Given

Compression Curve

Note that the maximum gain that can be entered in the table is 2.0

(minus 1 LSB). If more gain is required, the entire compression

curve may be shifted upward by using the post-compression gain

block following the compressor/limiter.

The AD1953 compressor/limiter also includes a look-ahead

compression feature. The idea behind look-ahead compression

is to prevent compressor overshoots by applying some digital

delay to the signal before the gain-control multiplier, but not to

the detector path. In this way, the detector can acquire the new

amplitude of the input signal before the signal actually reaches

the multiplier. A comparison of a tone burst fed to a conventional

compressor versus a look-ahead compressor is shown in Figure 12.

CONVENTIONAL COMPRESSOR GAIN

LOOK-AHEAD COMPRESSOR GAIN

HOLD TIME

Figure 12. Conventional Compression vs. Look-Ahead

Compression

REV. A

AD1953

–17–

In the look-ahead compressor, the gain has already been reduced

by the time the tone-burst signal arrives at the multiplier input.

Note that when using a look-ahead compressor, it is impor-

tant to set the detector hold time to a value that is at least the

same as the look-ahead delay time, or else the compressor

release will start too soon, resulting in an expanded “tail” of a

tone burst signal. The complete flow of the left/right dynamics

processor is shown in Figure 13.

LOOK-UP

TABLE

LINEAR

INTERPOLATION

MODIFIED RMS

DETECTOR WITH

LOG OUTPUT

HIGH BITS (1LSB = 3dB)

LOW BITS

TIME

CONSTANT

HOLD

RELEASE

DELAY

SPI-PROGRAMMABLE

LOOK-AHEAD DELAY

POST-COMPRESSION

GAIN, SPI-

PROGRAMMABLE

UP TO 30dB

(L+R)

Figure 13. Complete Dynamics Flow, Main Channels

The detector path works from a sum of left and right channels

((L+R)/2). This is the normal way that compressors are built,

and it counts on the fact that the main instruments in any stereo

mix are seldom recorded deliberately out of phase, especially in

the lower frequencies, which tend to dominate the energy spectrum

of real music.

The compressor is followed by a block known as post-compression

gain. Most compressors are used to reduce the dynamic range of

music by lowering the gain during loud signal passages. This

results in an overall loss of volume. This loss can be made up by

introducing gain after the compressor. In the AD1953, the

coefficient format used is 2.20, which has a maximum floating-

point representation of slightly less than 2.0. This means the

maximum gain that can be achieved in a single instruction is 6 dB.

To get more gain, the program in the AD1953 uses a cascade

of five multipliers to achieve up to 30 dB of post-compression gain.

To program the compressor/limiter, the following formulas may

be used to determine the 22-bit numbers (in 2.20 format) to be

entered into the parameter RAM.

RMS Time Constant

This can be best expressed by entering the time constant in

terms of dB/sec “raw” release rate (without the peak-riding circuit).

The attack rate is a rather complicated formula that depends on

the change in amplitude of the input sine wave.

rms tconst parameter

release rate

__ .–

⎛

⎝

⎜

⎞

⎠

⎟

10 10

10 0

where

rms_tconst_parameter = fractional number to enter into

the

SPI RAM (after converting to 22-bit 2.20 format)

release_rate = release rate of the raw rms detector in dB/sec. This

must be negative. f

= audio sampling rate.

RMS Hold Time

rms holdtime parameter f hold time

_ _ int _=×

()

where

rms_holdtime_parameter = integer number to enter into the SPI RAM

= audio sample rate

Hold_time = absolute time to wait before starting the release

ramp-down of the detector output

int() = integer part of expression

RMS Release Rate

rms decay parameter rms decay_ _ int _ / .=

()

1 096

Where rms_decay_parameter = decimal integer number to enter

into the SPI RAM

rms_decay = decay rate in dB/sec

int() = integer part of expression

Look-Ahead Delay

Lookahead delay parameter Lookahead delay f

__ _=×

Where Lookahead_delay = predictive compressor delay in abso-

lute time

= audio sample rate

The maximum Lookahead_delay_parameter value is 100.

Post-Compression Gain

Post compression gain parameter

Post compression gain linear

___

()

∧

Where Post_compression gain_linear is the linear post-compression

gain

^ = raise to the power

Subwoofer Compressor/Limiter

The subwoofer compressor/limiter differs from the left/right

compressor in the following ways:

1. The subwoofer compressor operates on a weighted sum of left

and right inputs (aa × Left + bb × Right), where aa and bb are

both programmable.

2. The detector input has a biquad filter in series with the input

in order to implement frequency-dependent compression

thresholds.

3. There is no predictive compression, as presumably the input

signals are filtered to pass only low frequencies, and therefore

transient overshoots are not a problem.

The subwoofer compressor signal flow is shown in Figure 14.

LOOK-UP

TABLE

LINEAR

INTERPOLATION

MODIFIED RMS

DETECTOR WITH

LOG OUTPUT

HIGH BITS (1LSB = 3dB)

LOW BITS

TIME

CONSTANT

HOLD

RELEASE

_SUB = K1  LEFT_IN + K2  RIGHT_IN

POST-COMPRESSION

GAIN, SPI-

PROGRAMMABLE

UP TO 30dB

BIQUAD

FILTER

Figure 14. Signal Flow for Subwoofer Compressor

The biquad filter before the detector can be used to implement

a frequency-dependent compression threshold. For example,

assume that the overload point of the woofer is strongly fre-

quency-dependent. In this case, one would have to set the

compressor threshold to a value that corresponded to the most

sensitive overload frequency of the woofer. If the input signal

happened to be mostly in a frequency range where the woofer

REV. A

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AD1953YSTZRL

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Analog Devices Inc.

Description:

Audio DSPs IC Digital Audio Processor

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AD1953YSTZRL

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