13823761625

Technology

Oversampled interpolation DAC
Date:June 18, 2025    Views:5

intro
    Oversampling and digital filtering help reduce the need for an anti-aliasing filter in front of the ADC. Reconstructed Dacs can apply the principles of oversampling and interpolation in a similar way. For example, digital audio CD players often use oversampling, where the basic data from the CD is updated at a rate of 44.1 kSPS. Early CD players used traditional binary Dacs and inserted "0" into parallel data, thereby increasing the effective update rate to 4, 8, or 16 times the base throughput rate. 4×, 8×, or 16× data flows through a digital interpolation filter, producing additional data points. The high oversampling rate moves the image frequency to a higher position, allowing simpler, lower cost filters with wider transition bands to be used. In addition, the SNR within the signal bandwidth is also increased due to the presence of processing gain. The Delta-Sigma DAC architecture extends this principle to the extreme by using a much higher oversampling rate, making it popular in modern CD players.
    The same principles of oversampling and interpolation can also be applied to high-speed Dacs in the field of communications, in order to reduce the requirements on the output filter and increase the SNR with the processing gain.


Reconstruct the DAC output spectrum
    The output of the reconstructed DAC can be represented as a series of rectangular pulses with a width equal to the inverse of the clock rate, as shown in Figure 1.


Figure 1: Unfiltered DAC output showing image and sin (x)/x roll down
    Note that at Nyquist frequency fc/2, the reconstructed signal amplitude is reduced by 3.92 dB. If desired, an anti-SIN (x)/x filter can be used to compensate for this effect. The mirror image of the fundamental signal appears as a result of the sampling function and is also attenuated by the sin(x)/x function.

Oversampled interpolation DAC
    The basic principle of oversampling/interpolating DAC is shown in Figure 2. N-bit input data words are received at rate fc. The digital interpolation filter operates at a clock rate equal to the oversampling frequency Kfc and inserts additional data points. The effect on the output spectrum is shown in Figure 2. At the Nyquist sampling frequency (A), the requirements for an analog antimirror filter can be quite high. Through oversampling and interpolation, the requirements for this filter can be greatly reduced, as shown in (B). In addition, the quantization noise is distributed over an area with a wider bandwidth than the original signal, so the signal-to-noise ratio is also improved. When the original sampling rate is doubled (K = 2), SNR is increased by 3 dB. For K = 4, SNR increases by 6 dB. Early CD players took advantage of this and were generally able to refine the algorithms in digital filters to more than N bits. Today, most Dacs in CD players are of the Delta-Sigma type.
    The earliest literature on the principle of oversampled/interpolated DAC is the 1974 paper by Ritchie, Candy, and Ninke (Ref. 1), and the patent filed by Mussman and Korte in 1981 (date of filing) (Ref. 2).



Figure 2: Oversampled interpolation DAC
    The following example uses some actual values to illustrate the oversampling principle. Suppose a conventional DAC is driven at an input word rate of 30 MSPS (see Figure 3A) with a DAC output frequency of 10 MHz. The image frequency component at 30-10 = 20 MHz must be attenuated by an analog anti-aliasing filter whose transition band begins at 10 MHz and ends at 20 MHz. Assuming that the image frequency must be attenuated by 60 dB, in the transition band of 10 MHz to 20 MHz (an octave), the filter must change from a passband break frequency of 10 MHz to a stopband attenuation of 60 dB. The filter provides attenuation of about 6 dB/ octave per pole. Therefore, in order to provide the desired attenuation, at least 10 poles are required. The narrower the transition band, the more complex the filter.



Figure 3: Analog filter requirements for fo = 10 MHz: (A) fc = 30 MSPS, (B) fc = 60 MSPS
    Suppose we increase the DAC update rate to 60 MSPS and insert a "0" between each of the original data sampling points. Now, the parallel data stream is 60 MSPS, but we have to determine the value of the zero value data point, this is done by passing the 60 MSPS data with 0 added through the process by a digital interpolation filter, which computs the additional data points. The response curve of the digital filter at 2× oversampling frequency is shown in Figure 3B. The analog anti-aliasing filter transition region is now 10 MHz to 50 MHz (the first image appears at 2FC-FO = 60-10 = 50 MHz). The transition region is slightly larger than 2 octaves, indicating that a 5 - or 6-pole filter is sufficient.
    The AD9773/AD9775/AD9777(12-/14-/16-bit) series Transmitting Dacs (TxDAC®) are 2 x, 4 x, or 8 x selectionable oversampled interpolation dual-channel Dacs, for which a simplified block diagram is shown in Figure 4. These devices are capable of handling 12/14/16-bit input word rates of up to 160 MSPS and maximum output word rates of 400 MSPS. Assuming that the output frequency is 50 MHz, the input update rate is 160 MHz, and the oversampling ratio is 2, the image frequency appears at 320 MHz - 50 MHz = 270 MHz, so the transition band of the analog filter is 50 MHz to 270 MHz. If there is no 2x oversampling, the image frequency occurs at 160 MHz - 50 MHz = 110 MHz, and the filter transition band is 50 MHz to 110 MHz.

    It should also be noted that the oversampled interpolation DAC supports a lower input clock rate and input data rate, so it is much less likely to generate noise within the system.


Σ-Δ typeDAC
    The principle of operation of a sigma-δ DAC is very similar to that of a Sigma-δ ADC, but in a Sigma-δ DAC, the noise shaping function is implemented using a digital modulator rather than an analog modulator.
    Unlike sigma-δ ADCs, Sigma-δ Dacs are mostly digital (see Figure 5A). It consists of an "interpolation filter" (a digital circuit that accepts data at a low rate, inserts 0 at a high rate, then applies the digital filter algorithm and outputs the data at a high rate), a Delta-type modulator (which is a low-pass filter for the signal and a high-pass filter for quantized noise, and converts the resulting data into a high-speed bitstream), and a 1-bit DAC. The DAC's output switches between the equivalent positive and negative reference voltages. The output is filtered in an external analog low-pass filter (LPF). Due to the high oversampling frequency, the complexity of the LPF is much lower than that of the traditional Nyquist sampling frequency.

Figure 5: Delta-Sigma DAC
    Sigma-δ Dacs can use a bit, which is the "multibit" architecture shown in Figure 5B, which is similar in principle to the interpolation Dacs discussed earlier, but with the addition of a Sigma-δ digital modulator.
    In the past, multibit Dacs were difficult to design due to the precision requirements of an N-bit internal DAC (although it only has n bits, it must have the linearity of the final n bits). However, the AD195x series audio Dacs solve this problem by leveraging proprietary "data scrambling" technology known as "data directed scrambling" to deliver outstanding performance across all audio specifications.
    Figure 6 shows the AD1955 multi-bit sigma-δ audio DAC. The AD1955 also uses data-directed scrambling technology, supports a variety of DVD audio formats, and has a very flexible serial port. The typical value of THD + N is 110 dB.

Figure 6: AD1955 multibit sigma-δ audio DAC

Sum up
    In modern data sampling systems, oversampling combined with digital filtering is a powerful tool. We have seen that the same basic principle applies to both ADCs and refactoring Dacs. The main advantage is that the requirements for anti-aliasing/anti-mirroring filters are reduced, and another advantage is that the SNR is increased due to the processing gain.
    The sigma-δ-type ADC and DAC architectures are terminal extensions of the oversampling principle and are the architecture of choice for most voice band and audio signal processing data converter applications.




    免责声明: 本文章转自其它平台,并不代表本站观点及立场。若有侵权或异议,请联系我们删除。谢谢!

    Disclaimer: This article is reproduced from other platforms and does not represent the views or positions of this website. If there is any infringement or objection, please contact us to delete it. thank you!
    矽源特科技ChipSourceTek

Copyright © 2017 copyright © 2017 ShenZhen ChipSourceTek Technology Co., Ltd. All Rights ReservedAll Rights Reserved 粤ICP备17060179号