AD1953_15 datasheet(14/36 Pages) AD

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AD1953

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first seven biquads would be used for speaker equalization and/

or tone controls, and the remaining filters would be programmed

to function as crossover filters. Note that there is a common

equalization section used for both the main and sub channels,

followed by crossover filters. This arrangement prevents any

interaction from occurring between the crossover filters and the

equalization filters. One section of the biquad IIR filter is shown

in Figure 5.

This section implements the transfer function:

() =

+×

()

−×

()

––

–

The coefficients a1, a2, b0, b1, and b2 are all in twos comple-

ment 2.20 format with a range from –2 to (+2 – 1 LSB). The

negative sign on the a1 and a2 coefficients is the result of adding

both the feed-forward “b” terms as well as the feedback “a” terms.

Some digital filter packages automatically produce the correct

a1 and a2 coefficients for the topology of Figure 5, while others

assume a denominator of the form 1 + a1

× Z–1 + a2 × Z–1. In this

case, it may be necessary to invert the a1 and a2 terms for

proper operation.

The biquad structure shown in Figure 5 is coded using double-

precision math to avoid limit cycles from occurring when low

frequency filters are used. The coefficients are programmed by

writing to the appropriate location in the parameter RAM through

the SPI port (see Table VI). There are two possible scenarios

for controlling the biquad filters:

1. Dynamic Adjustment (for example, Bass/Treble control or Para-

metric Equalizer)

When using dynamic filter adjustment, it is highly recommended

that the user employ the safeload mechanism to avoid temporary

instability when the filters are dynamically updated. This can

occur if some, but not all, of the coefficients are updated to new

values when the DSP calculates the filter output. The operation

of the Safeload registers is detailed in the Options for Parameter

Updates section.

2. Setting Static EQ Curve after Power-Up

If many of the biquad filters need to be initialized after power-

up (for example, to implement a static speaker-correction

curve), the recommended procedure is to set the processor

shutdown bit, wait for the volume to ramp down (about 20 ms),

and then write directly to the parameter RAM in Burst

Mode. After the RAM is loaded, the shutdown bit can be

deasserted, causing the volume to ramp back up to the initial

value. This entire procedure is click-free and faster than using

the Safeload mechanism.

The datapaths of the AD1953 contain an extra two bits on top

of the 24 bits that are input to the serial port. This allows up to

12 dB of boost without clipping. However, it is important to

remember that it is possible to design a filter that has less than

12 dB of gain at the final filter output, but more than 12 dB of

gain at the output of one or more intermediate biquad filter

sections. For this reason, it is important to cascade the filter

sections in the correct order, putting the sections with the

largest peak gains at the end of the chain rather than at the

beginning. This is standard practice when coding IIR filters and

is covered in basic books on DSP coding.

If gains larger than 12 dB cannot be avoided, then the coeffi-

cients b0 through b2 of the first biquad section may be scaled

down to fit the signal into the 12 dB maximum signal range, and

then scaled back up at the end of the filter chain.

Volume

Eight separate SPI registers are available to control the volume.

Three registers are used by the on-board program—one each for

the Left, Right, and Sub channels. These registers are special in

that they include automatic digital ramp circuitry for clickless

volume adjustment. The volume control word is in 2.20 format,

and gains from +2.0 to –2.0 are possible. The default value is

1.0. It takes 1024 audio frames to adjust the volume from 2.0

down to 0; in the normal case where the max volume is set to

1.0, it will take 512 audio frames for this ramp to reach zero.

Note that a Mute command is the same as setting the volume to

zero, except that when the part is unmuted, the volume returns to

its original value. These volume ramp times assume that the

AD1953 is set for the fast volume ramp speed. If the slow

setting is selected, it will take 8192 audio frames to reach zero

from a setting of 2.0. Correspondingly, it will take 4096 frames

to reach 0 volume from the normal setting of 1.0.

The volume blocks are placed after the biquad filter sections to

maximize the level of the signal that is passed through the filter

sections. In a typical situation, the nominal volume setting might be

–15 dB, allowing a substantial increase in volume when the user

increases the volume. The AD1953 was designed with an analog

dynamic range of > 112 dB, so that in the typical situation with

the volume set to –15 dB, the signal-to-noise ratio at the output

will still exceed 97 dB. Greater output dynamic ranges are possible

if the compressor/limiter is used, as the post-compression gain

parameter can boost the signal back up to a higher level. In this

case, the compressor will prevent the output from clipping when

the volume is turned up and the input signal is large.

Stereo Image Expander

The image-enhancement processing is based on ADI’s patented

Phat Stereo algorithm. The block diagram is shown in Figure 6.

1kHz

FIRST-ORDER LPF

LEVEL

LEFT IN

RIGHT IN

LEFT OUT

RIGHT OUT

–

Figure 6. Stereo Image Expander

The algorithm works by increasing the phase shift for low

frequency signals that are panned left or right in the stereo mix.

Since the ear is responsive to interaural phase shifts below 1 kHz,

this increase in phase shifts results in a widening of the stereo

image. Note that signals panned to the center are not processed,

resulting in a more natural sound. There are two parameters

that control the Phat Stereo algorithm: the Level variable,

which controls how much out-of-phase information is added to

the left and right channels, and the cutoff frequency of the

first-order low-pass filter, which determines the frequency range

of the added out-of-phase signals. For best results, the cutoff

frequency should be in the range of 500 Hz to 2 kHz. These

parameters are controlled by altering the parameter RAM locations

that store the parameters spread_level and alpha_spread.

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