Telephone calls clear as a bell
Smartphones can do almost everything you want, but their poor voice quality is still a vexing issue. Fraunhofer researchers have helped develop a new codec that makes voice quality as natural as if the person you’re calling is standing right next to you.
That’s because, for the first time, the entire audible frequency spectrum is transmitted. The new Enhanced Voice Services (EVS) standard promises a step change comparable with the transition from analogue CRT to digital flat-screen TVs. Specifications for standards of this type are extremely demanding.
“First of all, the codec must be capable of transmitting high-quality speech signals at relatively low data rates—so as not to compromise cost-efficiency,” says Dipl.-Ing Markus Multrus, who coordinated the software development part of the project at Fraunhofer IIS. Another requirement is that the codec should be sufficiently robust to recover from transmission errors, thereby ensuring that calls are not dropped due to poor reception.
Moreover, the codec must also be able to deliver similarly high quality when processing other types of signal, such as music on hold. This challenge is anything but simple, given that speech coding and audio coding are two separate worlds. The new codec therefore analyses the flow of signals every 20 milliseconds to distinguish between voice and music transmission, enabling appropriate algorithms to be applied.
Explains Dr Guillaume Fuchs, the research scientist who led the development of EVS at Fraunhofer, “The frequency range of the audio signals transmitted by currently available codecs only extends to 3.4kHz. Any frequencies above that limit are simply cut off, which is why phone calls sound so muffled. The new codec allows frequencies of up to 16 or even 20kHz to be transmitted, depending on the bit rate of the connection.”
That’s because, for the first time, the entire audible frequency spectrum is transmitted. The new Enhanced Voice Services (EVS) standard promises a step change comparable with the transition from analogue CRT to digital flat-screen TVs. Specifications for standards of this type are extremely demanding.
“First of all, the codec must be capable of transmitting high-quality speech signals at relatively low data rates—so as not to compromise cost-efficiency,” says Dipl.-Ing Markus Multrus, who coordinated the software development part of the project at Fraunhofer IIS. Another requirement is that the codec should be sufficiently robust to recover from transmission errors, thereby ensuring that calls are not dropped due to poor reception.
Moreover, the codec must also be able to deliver similarly high quality when processing other types of signal, such as music on hold. This challenge is anything but simple, given that speech coding and audio coding are two separate worlds. The new codec therefore analyses the flow of signals every 20 milliseconds to distinguish between voice and music transmission, enabling appropriate algorithms to be applied.
Explains Dr Guillaume Fuchs, the research scientist who led the development of EVS at Fraunhofer, “The frequency range of the audio signals transmitted by currently available codecs only extends to 3.4kHz. Any frequencies above that limit are simply cut off, which is why phone calls sound so muffled. The new codec allows frequencies of up to 16 or even 20kHz to be transmitted, depending on the bit rate of the connection.”

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