参数资料
型号: AD1954YSTZRL
厂商: Analog Devices Inc
文件页数: 7/36页
文件大小: 0K
描述: IC DAC AUDIO 3CHAN 26BIT 48LQFP
产品培训模块: Data Converter Fundamentals
DAC Architectures
标准包装: 2,000
系列: SigmaDSP®
位数: 26
数据接口: 串行
转换器数目: 3
电压电源: 模拟和数字
功率耗散(最大): 510mW
工作温度: -40°C ~ 105°C
安装类型: 表面贴装
封装/外壳: 48-LQFP
供应商设备封装: 48-LQFP(7x7)
包装: 带卷 (TR)
输出数目和类型: 6 电压,单极
采样率(每秒): 48k
AD1954
–15–
is related to the cutoff frequency of the first order low-pass filter
by the equation:
Alpha_Spread
– EXP
p Spread_Freq
fS
=
× ×
1.0
–2.0
where EXP is the exponential operator,
EXP
Spread_Freq is the low-pass
Spread_Freq
cutoff in Hz, and fS
ff is the audio sampling rate.
SS
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 sub-
woofer speakers is quite high (>500 Hz). In such a case, the stereo
spreading algorithm will not be effective, since the frequencies
that contribute to the spreading effect will come 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 chan-
nels.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 the 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 AD1954 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 fre-
quency 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
AD1954 is being operated in two-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 two-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 in-
troduced 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
NONLINEAR
CIRCUITS
VCA WITH EXP
CONTROL
THRESHOLD
SLOPE
OUT
Figure 7. Analog Compressor
The best analog compressors use rms detection as the signal
amplitude detector.The only class of detectors that is not sensi-
tive to the phase of the harmonics in a complex signal are rms
detectors.The ear also bases its loudness judgment on the overall
signal power and therefore using an rms detector results in the
best audible performance. Compressors that are based on peak
detection, while good for preventing clipping, are generally quite
poor for 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
In the case of a fast-responding rms detector, the detector envelope
will have a signal component in addition to the desired dc com-
ponent.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 AD1954 uses a modified rms algorithm to improve the relation-
ship 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.This figure shows two envelopes. One has the
harmonic distortion, as seen in the previous figure, and the other,
flatter envelope is the one produced by the AD1954.
REV. A
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