A range of voice codecs in a single chip

20th April 2016

Joe Bush

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Datasheet

The release of the new CMX7261 Function Image from CML Microcircuits also incorporates the G.723.1 digital voice coding algorithm of VoIP applications. The CMX7261 provides an end to end function with dual channel encoding, decoding, transcoding with no external DSP or DSP skills required. The device is well suited to software defined radio (SDR), voice over IP (VoIP) applications, wireless private branch exchange (PBX) and digital voice interconnect systems.

Two Function Images covering Linear PCM, G.711, G.729A and CVSD algorithms and Linear PCM, G.711 and G.723.1 algorithms are available, increasing the flexibility of the device. It also has the ability to deliver speech for a large number of applications without sacrificing performance.

Providing complete analogue to digital, digital to analogue or digital to digital conversion, the CMX7261 guarantees quality audio and ensures a short equipment design cycle. The device is built on CML’s FirmASIC technology, providing a platform for potential future enhancements.

Other key features include voice activity detection, C-BUS (SPI compatible) host serial control/data interface, multiple choice of input and output sources, streaming transfers, GPIO and encoder/decoder test mode support that allows verification of bit exactness and high audio quality for supported codecs.

Evaluation support is available via the PE0601-7261 EvKit. The CMX7261 operates from a 3.3V supply and includes selectable power saving modes and is available in small 64-pin VQFN/LQFP packaging.