Callers can come out of the cupboard with Wolfson’s audio codec for VoIP devices

Edinburgh, UK — Wolfson Microelectronics plc's WM8510 mono compressor/decompressor (codec) is the company's first product specifically designed for Voice over Internet Protocol (VoIP) devices and digital telephones.

The WM8510 delivers high-quality audio performance at low cost. A low power, highly integrated device, the WM8510 significantly reduces design costs and external component requirements.

According to Juniper Research, VoIP will account for over 12% of all telephony revenues by 2009 as VoIP evolves from being a replacement service for the public switch telephone network (PSTN) to a new converged service for the home and business user.

“The WM8510 delivers the performance and features that the fast-growing VoIP market demands,” said John Crawford, product manager, Wolfson Microelectronics. “It is designed specifically for digital telephone applications such as desktop phones, conference speakerphones and mobile telephone hands-free kits.”

The WM8510 includes a driver for a speaker or headset, two separate microphone inputs (e.g. handset and speakerphone) and two dedicated analog audio outputs with volume control. Advanced sigma-delta converters are used along with digital decimation and interpolation filters to give high quality audio at sample rates from 8 to 48 KS/sec.

Additional digital filtering options are available in the A/D converter path, as is an advanced automatic level control (ALC) function with noise gate. An on-chip phase-locked loop (PLL) is provided to generate the required master clock from an external reference clock. The PLL clock can also be output if required elsewhere in the system.

The WM8510 has a standard audio interface supporting transmission of audio data to and from the chip. The interface supports a number of audio data formats including -law and -law, PCM, I²S, and DSP mode, and can operate in master or slave modes.

The WM8510 operates at supply voltages from 2.5 V to 3.6 V, although the digital core can operate at voltages down to 1.62 V to save power. The speaker and mono outputs use a separate supply of up to 5 V, which enables increased output power if required. Different sections of the chip can also be powered down under software control by way of the selectable two or three-wire control interface.

The WM8510 is available in a 28-pin SSOP package and priced at $1.25 in 10,000-piece quantities.

For more information about Wolfson Microelectronics products in the U.S. call: (858) 676-5090 5090 or visit:

Wolfson's WM8510 audio codec offers VoIP telephony manufacturers better audio quality, smaller size and many more features than competing parts, according to Julian Hayes, Wolfson's vice president of marketing. “Codecs like this should begin to put an end to the low-voice quality we sometimes associate with phones,” he said.

Within the device the signal-to-noise ratio (SNR) of the D/A converter is 98 dB and 90 dB on the A/D converter. This is a really high value for SNR compared to most phones since it can capture a huge range between the highest value it can pick up and the noise floor, Hayes said. On the A/D converter side, this means that the part captures more elements of the user's voice to sound exactly the same, he said.

Other features include a wind noise filter on the part's DSP core that overcomes breath noises, and a powerful 900-mW speaker driver. An automatic level control (ALC) circuit allows the device to be programmed to amplify voice to a predetermined level so it doesn't sound like the speaker is in a cupboard, Hayes said.

Traditionally, inside all telephones, there is a codec for digitizing speech and regurgitating it so it can be heard — these are typically called µ-law or A-law codecs, which is a form of compression that's used in the public telephone network. These codecs also typically contain a pulse code modulation (PCM) interface to allow a seamless interface to digital signal processors.

Today, there is a new generation of desktop telephones — VoIP — and these phones have different codec requirements. “Generally speaking, they are much higher quality phones,” Hayes said. Old codecs can't be used anymore because the interface to the telephone line is no longer required. There is a different interface requirement because VoIP connects to the Internet instead of a public telephone network. To do this, the phone requires a different digital interface. In most cases, this is an Ethernet interface, so the VoIP telephone has an Ethernet interface on one side and a high quality audio codec is needed to digitize voice on the other side.

Wolfson's part is designed to interface with the digital chip that is used in VoIP phones and to provide a high quality audio subsystem for speech processing. It is providing a bridge to the digital chips that are used in VoIP without comprising basic high-quality audio performance, Hayes said.

Most landline telephones have low quality codecs in them. People are attracted to VoIP because it offers the potential for much better quality without spending a lot more money to have it, Hayes said. People are attracted to it because they don't have to pay for the Internet connection — assuming the people they are talking to have an Internet connection too.

Networking companies like Cisco and 3Com are looking at VoIP, which may change the landscape of the entire phone market. Growth in the conference phone market is very high where offices are using VoIP because it's fully integrated with the Ethernet network infrastructure, Hayes said. These new phones are challenging traditional landline phone makers since the cost of adding VoIP networks to existing Ethernet infrastructure is considered to be lower than adding landline phone systems, he said.

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