EVS Codec: The high quality next generation speech/audio coding

When you are comparing the audio quality of a phone call with the sound of a movie playing on the TV, the muffled audio of the standard telephone conversation becomes evident to everyone. This is mainly due to limitations such as very low audio bandwidth and poor performance on non-speech signals.

However, recent developments in speech and audio coding now promise a significant quality boost in conversational services, providing the full audio bandwidth for a more natural experience, improved speech intelligibility and listening comfort. Follower of AMR-WB, the HD voice codec and standardized in September 2014, Enhanced Voice Service (EVS) codec is the first 3GPP communication codec providing super-wideband (SWB) audio bandwidth for improved speech quality already at 9.6kbps. The codec was jointly developed by companies, representing operators, terminal, infrastructure and chipset vendors, as well as leading speech and audio coding experts.

The EVS voice codec offers input and output sampling at 8, 16, 32 and 48 kHz. In order to optimize the perceptual coding quality, an integrated bandwidth detector automatically adapts to the actual bandwidth of the input signal, which may be lower than the bandwidth indicated to the codec. The ability to switch the bit rate at every 20 ms frame further allows the codec to easily adapt to changes in channel capacity. One of its other features includes the discontinuous transmission (DTX) promoting comfort noise generation (CNG) and voice activity detection (VAD) algorithms. Inactive speech is coded using DTX/CNG in the SC-VBR coding mode. The codec also contains the jitter buffer management (JBM) system to tackle the jitter, i.e. variation in the delay of the received packets. EVS also includes a channel-aware mode operating at 13.2 kbit/s for both WB and SWB audio which helps to achieve increased robustness during unfortunate channel conditions.

Another advantage is the codec’s interoperability mode which enables backward compatibility with AMR-WB. Also, an advanced error concealment mechanism mitigates the quality impact of channel errors resulting in lost packets. The key technology of the codec is a flexible switching scheme between specialized coding modes for speech and music signals. It allows uninterrupted switching between circuit switched networks and VoLTE when network conditions warrant a transition. The codec’s performance on other signals, like music or mixed content, is comparable to modern audio codecs. Its wider audio bandwidth provides benefits to telephone users such as additional naturalness to voice quality and also enabling high quality for generic audio. It also provides a significant opportunity for operators to help mobilize enterprises by offering enterprise communication service packages featuring high-quality voice and videoconferencing, voice-recognition services and other similar services.

The feature set accompanied by unmatched speech and audio quality make the EVS voice codec, the most efficient and versatile codec for high quality communication in any type of network, in particular speech over WiFi and cellular LTE networks. EVS codec delivers a totally new experience to users everywhere, providing an audio quality that is close to transparency even for mobile communication services. The imminent introduction of the EVS voice codec (http://www.codecpro.com/codecsEVS.html) will therefore bring lasting benefits for both mobile operators and their customers.

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