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ADAV803ASTZ 数据表(PDF) 20 Page - Analog Devices |
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ADAV803ASTZ 数据表(HTML) 20 Page - Analog Devices |
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20 / 56 page ![]() ADAV803 Rev. 0 | Page 20 of 56 The s would appear for fS_IN equal to z = −125.1 dB + 96 kHz = −125.1 dB b ro. further reduction can be realize because only one interpolated sample is ta the of riod with a clock - mber of coefficients in ROM, the SRC r y tput nd the length of the convolution is increased by a factor of (fS_IN/fS_OUT). This technique is supported by the Fourier transform property 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’s FIFO block adjusts the left and right input samples and stores them for the FIR filter’s convolution cycle. The fS_IN counter provides the write address to the FIFO block and the ramp input to the digital servo loop. The ROM stores the coefficients for the FIR filter convolution 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 scaling o servo loop auto ati S_IN S_OUT sample rates tart of The worst-case images can be computed from the zero-order hold frequency response: maximum image = sin (× F/fS_INTERP)/(× F/fS_INTERP) 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. following worst-case image 192 kHz: Image at fS_INTERP − 96 kH Image at 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 y ze A d, ken 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 fS_OUT. Because there are 220 possible convolutions per fS_OUT period, arrival of the fS_OUT clock must be measured with an accuracy 1/201.3 GHz = 4.96 ps. Measuring the fS_OUT pe 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 polyphase coefficients for each tap, which requires a total of 226 coeffi cients. To reduce the nu 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 the convolution must be scaled. As the input sample rate rises ove the output sample rate, the antialiasing filter’s cutoff frequenc must be lowered, because the Nyquist frequency of the ou 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 a f the FIR filter length as well as the input data. The digital m cally tracks the f and f and provides the RAM and ROM start addresses for the s the FIR filter convolution. RIGHT DATA IN LEFT DATA IN FIFO DIGITAL SERVO LOOP fS_IN COUNTER ROM A ROM B ROM C ROM D SAMPLE RATE RATIO HIGH ORDER INTERP FIR FILTER L/R DATA OUT fS_IN fS_OUT SAMPLE RATE RATIO EXTERNAL RATIO 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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