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AD1954YSTZ 参数 Datasheet PDF下载

AD1954YSTZ图片预览
型号: AD1954YSTZ
PDF下载: 下载PDF文件 查看货源
内容描述: SigmaDSPâ ?? ¢ 3通道, 26比特信号处理DAC [SigmaDSP™ 3-Channel, 26-Bit Signal Processing DAC]
分类和应用: 消费电路商用集成电路
文件页数/大小: 36 页 / 1377 K
品牌: ADI [ ADI ]
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RMS Hol
d
T
ime  
In the look-ahead compressor, the gain has already been reduced  
by the time that the tone-burst signal arrives at the multiplier input.  
Note that when using a look-ahead compressor, it is important to  
set the detector hold time to a value that is at least the same as  
the look-ahead delay time or the compressor release will start too  
soon, resulting in an expanded tail of a tone-burst signal.The  
complete flow of the left/right dynamics processor is shown in  
Figure 13.  
rms_hold_time_parameter = int f × hold_time  
(
)
S
Where
rms_holdtime_parameter = the integer number to enter into  
the SPI RAM, fS = the audio sample rate, hold_time
= the abso-  
lute time to wait before starting the release ramp-down of the  
detector output, and
int
() = the integer part of the expression.  
RMS Release Rate  
rms_decay_parameter = int rms_decay /0.137  
(
)
DELAY  
POSTCOMPRESSION  
where
rms_decay_parameter = the decimal integer number to enter  
into the SPI RAM, rms_decay = the decay rate in dB/sec, and  
int
() = the integer part of the expression.  
SPI-PROGRAMMABLE  
LOOK-AHEAD DELAY  
GAIN, SPI-  
PROGRAMMABLE  
UP TO 30dB  
DELAY  
(L+R)  
2
Look-Ahead Delay  
lookahead_delay_parameter = lookahead_delay × fS  
HIGH BITS (1LSB = 3dB)  
MODIFIED RMS  
DETECTOR WITH  
LOG OUTPUT  
where
lookahead_delay = the predictive compressor delay in  
absolute time, fS = the audio sample rate, and the maximum  
lookahead_delay_parameter
value is 100.  
LOOK-UP  
TABLE  
LINEAR  
INTERPOLATION  
LOW BITS  
RELEASE  
HOLD  
TIME  
CONSTANT  
Postcompression Gain  
post_compression_gain_parameter =  
Figure 13. Complete Dynamics Flow, Main Channels  
post_compression_gain_linear1/5  
(
)
The detector path works from the sum of the left and right channels  
((L + R)/2).This is the normal way that compressors are built and  
counts on the fact that the main instruments in any stereo mix are  
seldom recorded deliberately out of phase, especially in the lower  
frequencies that tend to dominate the energy spectrum of real music.  
where
post_compression_gain_linear is the linear post-compression  
gain and ^ = the raise to the power.  
Subwoofer Compressor/Limiter  
The subwoofer compressor/limiter differs from the left/right  
compressor in the following ways:  
The compressor is followed by a block known as post-compression  
gain. Most compressors are used to reduce the dynamic range  
of music by lowering the gain during loud signal passages.This  
results in an overall loss of volume.This loss can be made up by  
introducing gain after the compressor. In the AD1954, the coef-  
ficient format used is 2.20, which has a maximum floating-point  
representation of slightly less than 2.0.This means that the maxi-  
mum gain that can be achieved in a single instruction is 6 dB.To  
get more gain, the program in the AD1954 uses a cascade of five  
multipliers to achieve up to 30 dB of post-compression gain.  
1. The subwoofer compressor operates on a weighted sum of the  
left and right inputs (aa
Left + bb
Right), where aa and  
bb are both programmable.  
2. The detector input has a biquad filter in series with the input  
in order to implement frequency-dependent compression  
thresholds.  
3. There is no predictive compression since presumably the input  
signals are filtered to pass only low frequencies and therefore  
transient overshoots are not a problem.  
To program the compressor/limiter, the following formulas may  
be used to determine the 22-bit numbers (in 2.20 format) to be  
entered into the parameter RAM.  
The subwoofer compressor signal flow is shown in Figure 14.  
V
_SUB = k1 LEFT_IN + K2 RIGHT_IN  
IN  
RMSTime Constant  
This can be best expressed by entering the time constant in terms  
of dB/sec raw release rate (without the peak-riding circuit).The  
attack rate is a rather complicated formula that depends on the  
change in amplitude of the input sine wave.  
POSTCOMPRESSION  
GAIN, SPI-  
PROGRAMMABLE  
UP TO 30dB  
HIGH BITS (1LSB = 3dB)  
MODIFIED RMS  
DETECTOR WITH  
LOG OUTPUT  
BIQUAD  
FILTER  
LOOK-UP  
TABLE  
LINEAR  
INTERPOLATION  
release_rate  
LOW BITS  
10.0 × fS  
)
rms_tconst_parameter = 1.0 10(  
RELEASE  
HOLD  
TIME  
CONSTANT  
where
rms_tconst_parameter
= the fractional number to enter into  
the SPI RAM (after converting to 22-bit 2.20 format), and the  
release_rate = the
release rate
of the raw rms detector in dB/sec.  
This must be negative, and fS = the audio sample rate.  
Figure 14. Signal Flow for Subwoofer Compressor  
REV. A  
–17–  
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