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ADAV801ASTZ Datasheet(PDF) 20 Page - Analog Devices

Part # ADAV801ASTZ
Description  Audio Codec for Recordable DVD
PDF  56 Pages
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Manufacturer  AD [Analog Devices]
Direct Link  http://www.analog.com
Logo AD - Analog Devices

ADAV801ASTZ Datasheet(HTML) 20 Page - Analog Devices

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ADAV801
Rev. 0 | Page 20 of 56
The
mputed from the zero-order
× F/fS_INTERP)/(× F/fS_INTERP)
Hardware Model
The output rate of the low-pass filter in Figure 30 is the
interpolation rate:
220 × 192,000 kHz = 201.3 GHz
Sampling at a rate of 201.3 GHz is clearly impractical, not to
mention the number of taps required to calculate each
interpolated sample. However, because interpolation by 220
involves zero-stuffing 220−1 samples between each fS_IN sample,
most of the multiplies in the low-pass FIR filter are by zero. A
further reduction can be realized, because only one interpolated
sample is taken at the output at the fS_OUT rate, so only one
convolution needs to be performed per fS_OUT period instead of
220 convolutions. A 64-tap FIR filter for each fS_OUT sample is
sufficient to suppress the images caused by the interpolation.
One difficulty with the above approach is that the correct
interpolated sample must be selected upon the arrival of
UT
.
Because th
eriod, the
lyphase
26
,
e
aled. As the input sample rate rises over
the output sample rate, the antialiasing filter’s cutoff frequency
y
’s FIFO block adjusts the
left and right input samples and ores them for the FIR filter’s
c
to the FIFO blo
l servo loop.
tion
aling
tal
worst-case images can be co
hold frequency response:
maximum image = sin (
where:
F is the frequency of the worst-case image that would be
220 × fS_IN ± fS_IN/2.
fS_INTERP is fS_IN × 220.
The following worst-case images would appear for fS_IN equal to
192 kHz:
Image at fS_INTERP − 96 kHz = −125.1 dB
Image at fS_INTERP + 96 kHz = −125.1 dB
fS_O
ere are 220 possible convolutions per fS_OUT p
arrival of the fS_OUT clock must be measured with an accuracy of
1/201.3 GHz = 4.96 ps. Measuring the fS_OUT period with a clock
of 201.3 GHz frequency is clearly impossible; instead, several
coarse measurements of the fS_OUT clock period are made and
averaged over time.
Another difficulty with the above approach is the number of
coefficients required. Because there are 220 possible convolu-
tions with a 64-tap FIR filter, there must be 220 po
coefficients for each tap, which requires a total of 2 coeffi-
cients. To reduce the number of coefficients in ROM, the SRC
stores a small subset of coefficients and performs a high order
interpolation between the stored coefficients.
The above approach works when fS_OUT > fS_IN. However, when
the output sample rate, fS_OUT, is less than the input sample rate
fS_IN, the ROM starting address, input data, and length of th
convolution must be sc
must be lowered, because the Nyquist frequency of the output
samples is less than the Nyquist frequency of the input samples.
To move the cutoff frequency of the antialiasing filter, the
coefficients are dynamically altered and the length of the
convolution is increased by a factor of (fS_IN/fS_OUT).
This technique is supported by the Fourier transform propert
that, if f(t) is F(ω), then f(k × t) is F(ω/k). Thus, the range of
decimation is limited by the size of the RAM.
SRC Architecture
The architecture of the sample rate converter is shown in
Figure 32. The sample rate converter
st
onvolution cycle. The fS_IN counter provides the write address
ck and the ramp input to the digita
The ROM stores the coefficients for the FIR filter convolu
and performs a high order interpolation between the stored
coefficients. The sample rate ratio block measures the sample
rate for dynamically altering the ROM coefficients and sc
of the FIR filter length as well as the input data. The digi
servo loop automatically tracks the fS_IN and fS_OUT sample rates
and provides the RAM and ROM start addresses for the start of
the FIR filter convolution.
RIGHT DATA IN
LEFT DATA IN
FIFO
ROM A
ROM B
DIGITAL
SERVO LOOP
fS_IN
COUNTER
ROM C
ROM D
fS_IN
fS_OUT
SAMPLE RATE RATIO
SAMPLE
RATE RATIO
EXTERNAL
RATIO
INTERP
FIR FILTER
L/R DATA OUT
HIGH
ORDER
Figure 32. Architecture of the Sample Rate Converter
The FIFO receives the left and right input data and adjusts the
amplitude of the data for both the soft muting of the sample
rate converter and the scaling of the input data by the sample
rate ratio before storing the samples in the RAM. The input data
is scaled by the sample rate ratio, because, as the FIR filter
length of the convolution increases, so does the amplitude of the
convolution output. To keep the output of the FIR filter from
saturating, the input data is scaled down by multiplying it by
(fS_OUT/fS_IN) when fS_OUT < fS_IN. The FIFO also scales the input
data for muting and unmuting of the SRC.
The RAM in the FIFO is 512 words deep for both left and right
channels. An offset to the write address provided by the fS_IN
counter is added to prevent the RAM read pointer from
overlapping the write address. The minimum offset on the SRC
is 16 samples. However, the group delay and mute-in register
can be used to increase this offset.



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