AD1953YST Analog Devices Inc, AD1953YST Datasheet - Page 14

• Manufacturer Part Number: AD1953YST
• Description: IC DAC AUDIO 3CH 26BIT 48-LQFP
• Manufacturer: Analog Devices Inc
• Series: SigmaDSP®r

Specifications of AD1953YST

• Rohs Status: RoHS non-compliant
• Number Of Bits: 26
• Data Interface: Serial
• Number Of Converters: 3
• Voltage Supply Source: Analog and Digital
• Power Dissipation (max): 540mW
• Operating Temperature: -40°C ~ 105°C
• Mounting Type: Surface Mount
• Package / Case: 48-LQFP
• For Use With: EVAL-AD1953EBZ - BOARD EVAL FOR AD1953 3CH 24BIT
• Settling Time: -

AD1953
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
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-
2. Setting Static EQ Curve after Power-Up
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.
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.
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.
H Z
( )
=
(
(
b
1
0
−
+
a
b
1
1
×
×
Z
Z
–
–
1
1
–
+
a
b
2
2
×
×
Z
–1
Z
–
–
2
+ a2 × Z
2
)
)
–1
. In this
–14–
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.
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.
RIGHT IN
LEFT IN
Figure 6. Stereo Image Expander
+
–
LEVEL
FIRST-ORDER LPF
1kHz
–
+
RIGHT OUT
LEFT OUT
–
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