Using audio codecs IP as the digital audio hub in mobile multimedia systems
Mobile multimedia devices process and combine audio signals from a variety of sources, including the baseband processor, Bluetooth enabled devices, and WiFi networks. The result is that today's smartphones and tablets are 'digital audio hubs' that must receive multiple asynchronous digital audio signals, synchronize them, and output them on loudspeakers or headsets.
Tablet/smartphone system-on-chips (SoCs) designers are faced with the challenge of implementing the complex audio mixing functions in the most cost-effective way possible. Traditionally, this function can be implemented in the application processor or in a dedicated audio processor; however, this is not the most cost-effective way to use the limited processing resources available to the system.
By integrating an audio analog codec that implements the 'audio hub' functionality and is able to process and mix audio signals from asynchronous sources, system designers can free the scarce main processor resources for more relevant tasks and simplify the system design, thus achieving a more effective solution.
This article will analyze:
- The benefits of having the audio codec in mobile multimedia systems operating as a digital audio hub to interconnect the different audio signal sources and destinations, each having independent clock domains
- How to synchronize and combine the various audio streams originated by different sources in the system, using built-in asynchronous sample rate converters (ASRCs).
By leveraging the latest improvements in audio codec IP, designers and system architects will be able to deliver the tightly integrated solutions that will make their SoCs stand out from the competition while minimizing costs.
Audio Codec Requirements
The core of an audio codec is composed of two types of data converters: an analog-to-digital converter (ADC) for recording and a digital-to-analog converter (DAC) for playback. For a stereo or multi-channel codec, these are replicated accordingly. Figure 1 shows a typical block diagram of a stereo audio codec.
In the analog side, the record channel includes amplifiers with volume controls to bring both the weak microphone levels and the large interconnect line levels to the input range of the ADC. The playback channel includes output drivers able to directly connect to earphones or to small speakers, each with its respective volume controls. There is also a low-noise power supply for microphone biasing.
The digital side includes 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 ensures that all multi-rate blocks are synchronized with each other and support the required sampling rate combinations.