AD1954 Analog Devices, AD1954 Datasheet - Page 14

• Manufacturer Part Number: AD1954
• Description: SigmaDSP Digital Audio Processor
• Manufacturer: Analog Devices

AD1954
The datapaths of the AD1954 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
Three separate SPI registers are used to control the volume—
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 therefore 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 a set-
ting the volume to zero, except that when the part is un-muted,
the volume returns to its original value.
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 AD1954 was designed with an analog
dynamic range of > 115 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 100 dB. Greater output dynamic ranges are pos-
sible 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 below.
“Burst Mode.” After the RAM is loaded, the shutdown bit
can be de-asserted, causing the volume to ramp back up to
the initial value. This entire procedure is click-free, and faster
than using the “Safeload” mechanism.
RIGHT IN
LEFT IN
+
–
LEVEL
FIRST-ORDER LPF
PRELIMINARY TECHNICAL DATA
1 kHz
–
+
RIGHT OUT
LEFT OUT
–
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 inter-aural 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 lowpass 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.”
The “spread_level” is a linear number in 2.20 format that multi-
plies 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:
where EXP is the exponential operator, spread_freq is the low-pass
cutoff in Hz, and f
Note that the stereo spreading algorithm assumes that frequen-
cies 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 frequen-
cies 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 placements 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 AD1954 is quite sophisticated, and
is comparable in many ways to professional compressor/limiters
used in the pro audio and broadcast fields. It uses RMS/peak
detection with adjustable attack/hold/release, look-ahead com-
pression, and table-based entry of the input/output curve for
complete flexibility.
Alpha spread
_
=
S
1 0
is the audio sampling rate.
. –
EXP



– .
2 0
×
π
×
spread freq
f
S
_


