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ADAV801ASTZ 数据表(PDF) 21 Page - Analog Devices |
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ADAV801ASTZ 数据表(HTML) 21 Page - Analog Devices |
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21 / 56 page ![]() ADAV801 Rev. 0 | Page 21 of 56 ut samples added to the write pointer of the (512 − 16)/64 taps = 7.75 for short group delay and (512 − 64)/64 taps = 7 for long group delay. o of fS_IN/fS_OUT The number of inp FIFO on the SRC is 16 plus Bit 6 to Bit 0 of the group delay register. This feature is useful in varispeed applications to prevent the read pointer to the FIFO from running ahead of the write pointer. When set, Bit 7 of the group delay and mute-in register soft-mutes the sample rate. Increasing the offset of the write address pointer is useful for applications in which small changes in the sample rate ratio between fS_IN and fS_OUT are expected. The maximum decimation rate can be calculated from the RAM word depth and the group delay as The digital servo loop is essentially a ramp filter that provides the initial pointer to the address in RAM and ROM for the start of the FIR convolution. The RAM pointer is the integer output of the ramp filter, and the ROM is the fractional part. The digital servo loop must provide excellent rejection of jitter on the fS_IN and fS_OUT clocks, as well as measure the arrival of the fS_OUT clock within 4.97 ps. The digital servo loop also divides the fractional part of the ramp output by the rati to dynamically alter the ROM coefficients when fS_IN > fS_OUT. REG 0x00 BITS 1–0 REG 0x76 BIT 0 REG 0x76 BIT 1 REG 0x62 BITS 7–6 DIR PLAYBACK AUXILIARY IN ADC SRC MCLK SRC OUTPUT SRC SRC INPUT Figure 33. Clock and Datapath Control on the SCR The digital servo loop is implemented with a multirate filter. To ttle the digital servo loop filter more quickly upon startup or a added to the ample rate is easonable value, the digital During fast mode, the MUTE_OUT bi e error e he u at clicks ght pres l audio tput of e mu Bit 7 of he group delay and r l t ged to he MU an be n inter when the SRC m h ing sample rate onve ly. e /fS_IN) × 220 ratio for se change in the sample rate, a fast mode has been filter. When the digital servo loop starts up or the s changed, the digital servo loop enters fast mode to adjust and settle on the new sample rate. Upon sensing that the digital servo loop is settling down to a r servo loop returns to normal (or slow) mode. t in the sample rat regist r is asserted to let t ser know th or pops mi be ent in the digita data. The ou the SRC can b ted by asserting t mute registe unti he SRC has chan slow mode. T TE_OUT bit c set to generate a rupt changes to slow ode, indicating that t e data is be c rted accurate The frequency responses of the digital servo loop for fast mod and slow mode are shown in Figure 34. The FIR filter is a 64-tap filter when fS_OUT ≥ fS_IN and is (fS_IN/fS_OUT) × 64 taps when fS_IN > fS_OUT. The FIR filter performs its convolution by loading in the starting address of the RAM address pointer and the ROM address pointer from the digital servo loop at the start of the fS_OUT period. The FIR filter then steps through the RAM by decrementing its address by 1 for each tap, and the ROM pointer increments its address by the (fS_OUT fS_IN > fS_OUT or 220 for fS_OUT ≥ fS_IN. Once the ROM address rolls over, the convolution is completed. FREQUENCY (Hz) 0 –20 –40 –60 –80 –100 –120 –140 –160 –180 –200 0.01 0.1 1 10 100 1k 10k 100k SLOW MODE FAST MODE –220 Figure 34. Frequency Response of the Digital Servo Loop. fS_IN is the X-Axis, fS_OUT = 192 KHz, Master Clock is 30 MHz The convolution is performed for both the left and right channels, and the multiply accumulate circuit used for the convolution is shared between the channels. The fS_IN/fS_OUT sample rate ratio circuit is used to dynam cally alter the in the ROM when fS_IN > fS_OUT. The ratio is calculated by comparing the output of an fS_OUT counter to the output of an fS_IN counter. If fS_OUT > fS_IN, the ratio is held at one. UT , the sample rate ratio is updated, if it is different by more than two fS_OUT periods from the previous fS_OUT to fS_IN comparison. This is done to provide some hysteresis to prevent i coefficients If fS_IN > fS_O the filter length from oscillating and causing distortion. |
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