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Managing different clock frequencies of audio codecs

Posted: 02 Dec 2011     Print Version  Bookmark and Share

Keywords:Hi-Fi  audio-video  digital filters  analogue-to-digital converter 

On a DAC, the image bands cause a stair-like waveform as shown in figure 2. The filter smoothes the waveform and reduces the high-frequency energy. If this high-frequency energy was not removed, it would waste power and cause intermodulation distortion in the output drivers, causing the loud-speakers to generate audible noises.

On an ADC, the filter removes any out-of-band noise picked up at the input or generated within the ADC, as shown in figure 3. If this is not removed when the signal is re-sampled at the standard audio rate, the noise would be folded down in-band due to aliasing and becomes audible.

Figure 2: The digital filter up-samples and smoothes the signal waveform before being applied to the DAC.

Figure 3: On an ADC, any out-of-band noises (red signal on the left diagram) would be folded down into the signal band when the sample rate is re-sampled to the standard audio rate at the output (diagram on the right).

Clocks and sampling rates
Digital audio signals are sampled at standard frequencies. Due to legacy from the old Redbook CD, many audio recordings use the standard 44.1 kS/s rate. This unconventional number is derived from an early practice of reusing PAL videotape equipment for audio recordings. Modern audio systems, like DVDs, use 48 kS/s and its multiples 96 kS/s and 192 kS/s.

Voice applications, such as those in cell phones, use 8 kS/s and its multiples, 16 kS/s and 32 kS/s. Some applications may also use multiples of 44.1 kS/s, namely 88.2 kS/s and 176.4 kS/s. Since the data converters must operate at oversampled frequencies, typically 128X, or 256X, the required master clock frequencies to drive the data converters would be in the range of 5 to 12MHz.

An audio codec must therefore support a wide variety of audio sample rates and accommodate a range of master clock frequencies facilitating its operation in the application. It is not a straightforward objective due to the multitude of combinations and restrictions in the possible clock frequency ratios. For this reason, the digital filters must include programming of its sample rate conversion.

For example, let's consider a practical case with an audio rate of 48 kS/s and the converter's sampling frequency of 12.288 MS/s. The resulting sample rate conversion is 256X. Now, for supporting 96 kS/s, the filters are reconfigured for a sample rate conversion of 128X. And for supporting 192 kS/s, the filters are reconfigured for a sample rate conversion of 64X. The sampling frequency of the data converters stays the same at 12.288 MS/s because the audio band limit is fixed at 20kHz. For the 44.1-kS/s audio rate family, the corresponding master clock would be 11.2896kHz.

By proper reconfiguration of the digital filters, sample rate conversion and flexible clock frequency choices, it is possible to support a wide range of audio-video applications with advanced audio codecs. There are several solutions for these applications that help designers understand the trade-offs needed to minimise the costs of their SoCs.

PLL for audio clock
Many applications such as portable products cannot have dedicated crystal oscillators for the audio codec, due to its space and/or cost limitations. The audio codec must be able to support the different audio rates from the available host master clock, which is often the USB clock operating at 12MHz or a multiple of it. In which case, a phase-locked loop (PLL) can be used to generate the required audio clocks. But this PLL is relatively complex due to the very fine frequency resolution required for supporting all the frequency combinations, while at the same time providing a low-jitter output clock for performance. Other solutions not requiring a PLL would be preferable.

PLL-less technique
An alternative solution is the PLL-less technique of re-using the USB clock and avoiding adding a dedicated PLL for audio. USB is a very popular interface and almost universally present in any application. Either 12MHz or 24MHz clocks are used and have relatively low jitter, which is an important requirement for audio. A USB clock of 12MHz can support the 48-kS/s audio rate because it is an integer multiple (12,000 = 250 x 48). To use it, the filters sample rate conversion needs to be reconfigured from the nominal 256X to 250X.

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