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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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AD1954  
L/R DYNAMICS PROCESSOR  
EQ AND CROSSOVER FILTERS  
8  
DELAY  
(0ms–2.3ms)  
OUT  
DAC  
DAC  
INTERPOLATION  
LEFT  
DELAY  
(0ms–3.7ms)  
CROSSOVER  
7 BIQUAD  
(2 FILTERS)  
FILTERS  
HPF/  
DEEMPH  
IN  
LEFT  
LEVEL DETECT,  
LOOK-UP TABLE  
DELAY  
(0ms–3.7ms)  
HPF/  
DEEMPH  
CROSSOVER  
(2 FILTERS)  
7 BIQUAD  
FILTERS  
IN  
RIGHT  
8  
DELAY  
(0ms–2.3ms)  
OUT  
RIGHT  
INTERPOLATION  
L/R REINJECTION  
LEVEL  
LEVEL DETECT,  
LOOK-UP TABLE  
1 BIQUAD  
FILTER  
SUB CHANNEL  
L/R MIX  
DELAY  
(0ms–3.7ms)  
SUBWOOFER  
OUTPUT  
CROSSOVER  
(3 FILTERS)  
MONO DAC  
SUB DYNAMICS PROCESSOR  
Figure 2. Signal Processing Flow  
SIGNAL PROCESSING  
Signal Processing Overview  
The AD1954 uses two different numeric formats: one for the  
coefficient values (stored in the parameter RAM) and one for the  
signal data values.The coefficient format is as follows:  
Figure 2 shows the signal processing flow diagram of the AD1954.  
The AD1954 is designed to provide all the signal processing  
functions commonly used in 2.0 or 2.1 playback systems. A seven-  
biquad equalizer operates on the stereo input signal.The output of  
this equalizer is fed to a two-biquad crossover filter for the main  
channels, and the mono sum of the left and right equalizer outputs  
is fed to a three-biquad crossover filter for the subchannel. Each  
of the three channels has independent delay compensation.There  
are two high quality compressor/limiters available: one operating  
on the left/right outputs and one operating on the subwoofer chan-  
nel.The subwoofer output may be blended back into the left/right  
outputs for 2.0 playback systems. In this configuration, the two  
independent compressor/limiters provide two-band compression,  
which significantly improves the sound quality of compressed  
audio. In addition, the main channels have a stereo widening  
algorithm that increases the perceived spread of the stereo image.  
Coefficient Format  
Coefficient Format: 2.20  
Range: –2.0 to +(2.0 – 1 LSB)  
Examples:  
1000000000000000000000 = –2.0  
1100000000000000000000 = –1.0  
1111111111111111111111 = (1 LSB below 0.0)  
0000000000000000000000 = 0.0  
0100000000000000000000 = 1.0  
0111111111111111111111 = (2.0 – 1 LSB)  
This format is used because standard biquad filters require  
coefficients that range between +2.0 and –2.0. It also allows gain  
to be inserted at various places in the signal path.  
Internal DSP Signal Data Format  
Input Data Format: 1.23  
This is sign extended when written to the data memory of the  
AD1954.  
Internal DSP Signal Data Format: 3.23  
Range: –4.0 to +(4.0 – 1 LSB)  
Examples:  
Most of the signal processing functions are coded using full 48-bit  
double-precision arithmetic.The input word length is 24 bits, with  
two extra headroom bits added in the processor to allow internal  
gains up to 12 dB without clipping (additional gains can be  
accommodated by scaling down the input signal in the first biquad  
filter section).  
10000000000000000000000000 = –4.0  
11000000000000000000000000 = –2.0  
11100000000000000000000000 = –1.0  
11111111111111111111111111 = (1 LSB below 0.0)  
00000000000000000000000000 = 0.0  
00100000000000000000000000 = 1.0  
01000000000000000000000000 = 2.0  
01111111111111111111111111 = (4.0 – 1 LSB).  
A graphical user interface (GUI) is available for evaluation of  
the AD1954 (Figure 3).This GUI controls all of the functions of  
the chip in a very straightforward and user friendly interface. No  
code needs to be written to use the GUI to control the chip. For  
more information on AD1954 software tools, send an e-mail to  
SigmaDSP@analog.com.  
Each section of this flow diagram will be explained in detail on  
the following pages.  
The sign extension between the serial port and the DSP core  
allows for up to 12 dB of gain in the signal path without internal  
clipping. Gains greater than 12 dB can be accommodated by  
scaling the input down in the first biquad filter and scaling the  
signal back up at the end of the biquad filter section.  
Numeric Formats  
It is common in DSP systems to use a standardized method of  
specifying numeric formats.To better comprehend issues relating to  
precision and overflow, it is helpful to think in terms of fractional  
twos complement number systems. Fractional number systems  
are specified by an A.B format, where A is the number of bits to  
the left of the decimal point, and B is the number of bits to the  
right of the decimal point. In a twos complement system, there is  
also an implied offset of one-half of the binary range; for example,  
in a twos complement 1.23 system, the legal signal range is  
–1.0 to +(1.0 – 1 LSB).  
A digital clipper circuit is used between the output of the DSP  
core and the input to the DAC
-
modulators to prevent over-  
loading the DAC circuitry (see Figure 4). Note that there is a gain  
factor of 0.75 used in the DAC interpolation filters, and therefore  
signal values of up to 1/0.75 will pass through the DSP without  
clipping. Since the DAC is designed to produce an analog output  
of 2V rms (differential) with a 0 dB digital input, signals between  
–12–  
REV. A  
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