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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 

Audio processing is important to many consumer electronic applications such as mobile phones, MP3 players and a host of other products. While size and power consumption are often critical SoC design requirements, the market also demands high-quality high fidelity (Hi-Fi) audio capabilities. To meet this consumer demand, designers are now embedding audio codecs into their next-generation, advanced SoCs.

The audio codec creates the interface between the digital host processor and the audio transducers, such as microphones and speakers. It is also responsible for several routine audio functions, thereby alleviating the workload on the host processor.

The clocks required by the data converters on an audio codec depend on the audio material sampling rates as well as on the clocks available on the host application and SoC. The combinations are quite complex due to the multitude of audio sample rate options and available host clocks. To further complicate matters, in audio-video (A/V) applications, the audio clocks need to also be synchronised with the video clocks required by the video data converters. Therefore, many designers are confronted with complex choices when deciding on trade-offs to minimise system costs related to clock generation and interfacing a multitude of sample rates.

The digital filters play an important role in synchronising the different clocks because they process the digital samples between the digital audio interface and the audio data converters, and therefore, can perform sampling rate conversions. This article will review the functions of digital filters in audio codecs and will provide several examples to illustrate how they can interface to a range of sample-rates and clock environments.

Audio processing
The audio codec is composed of two types of data converters: a digital-to-analogue converter (DAC) for playback and an analogue-to-digital converter (ADC) for recording.

On the digital side, there are multiple blocks. The most important are the digital audio filters that convert the data rate to the oversampled clocks of the data converters and remove the high-frequency noise outside the audio band. Also important is a clock management block, which makes sure that all multi-rate blocks are synchronised with each other and supports all the required sampling rate combinations.

Today, data converters in audio codecs operate at highly oversampled frequencies, which mean that their conversion frequency is much higher than the audio band, often by a factor of 100 or more. For example, assuming a Redbook CD player has an audio data rate of 44.1 kSamples per second (kS/s), the typical oversampling rate is 128X, leading to the DAC's conversion rate of 5.6448 MSamples per second (MS/s).

Figure 1: Audio signal sampled at FS and its spectrum replicas at 2FS, 3FS, (in orange).

Requirement for digital audio filters
The main reason filters are required on an audio codec is to remove the aliasing or imaging bands. These are replicas of the signal band around the multiples of the audio sampling rate (FS) and are a result of the multi-rate operation. For example, an audio stream at 44.1 kS/s up-sampled to 5.6448 MS/s has spectrum replicas of around 88.2kHz, 132.3kHz. This is a result of the Nyquist Sampling theorem, as illustrated in figure 1.

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