What is the next big thing in LTE-based 4G mobile networks? Apparently, it’s Voice over LTE (VoLTE) these days, especially after 3GPP has released the Enhanced Voice Services (EVS) codec that industry watchers call a breakthrough in audio and voice communications.
Long Term Evolution or LTE is the first cellular system that has been developed for data applications from grounds-up. So voice service has mostly been a dilemma for the LTE business. Initially, mobile operators began moving the voice calls to the 2G and 3G networks as a stop-gap measure. Then, in the early 2010s, mobile operators like MetroPCS and SK Telecom started launching VoLTE services that used both AMR narrowband and wideband codecs.
The 2G cellular networks, such as GSM and CDMA, mostly use adaptive multi-rate (AMR) codec that operates on narrowband 200-3400 Hz signals at variable bit rates in the range of 4.75 Kbps to 12.2 Kbps. However, the traditional voice standard—also known as AMR narrowband or AMR-NB—falls short in terms of voice clarity and noise cancellation because it sacrifices voice quality to enable lower bandwidths.
EVS is a major breakthrough for VoLTE service
The AMR wideband or AMR-WB standard improves speech quality and audio coding through a wider bandwidth of 50-7000 Hz and consumes relatively less channel capacity, from 12 kbps to nearly 24 Kbps. The AMR-WB codec—synonymous with HD voice—features enhanced audio processing, multiple microphones and speakers and improved echo cancellation to enhance voice quality and reduce background noise.
However, AMR-WB has taken too long for commercial realization, and the fact that devices on both ends are required to be HD voice-capable has led to limited availability. Then, in 2014, 3GPP finalized the EVS codec that goes well beyond AMR-WB in terms of speech quality, wider frequency range, and bandwidth utilization.
The Physics of EVS
The EVS voice codec offers HD quality voice of AMR-WB in less bandwidth than the AMR-WB codec. So, mobile operators like T-Mobile, now using 24 Kbps for HD voice, can employ super-wideband EVS and have the same audio quality with 5.95 Kbps to 7.36 Kbps. Furthermore, EVS enables innovative music and audio applications like live-to-air studio quality calls from mobile phones.
EVS uses 50 Hz to 14 KHz bandwidth that encompasses narrowband, wideband, super-wideband and full-band voice communications. It’s backward compatible with both AMR-NB and AMR-WB codec standards and can be used for even 2G and 3G networks to reduce bandwidth demands while maintaining the same voice quality. EVS has put in place error resilience mechanism for both circuit-switched 2G and 3G voice services as well as packet-switched Voice over IP (VoIP) applications.
EVS and the evolution of mobile voice
(Image credit: Qualcomm Inc.)
EVS, a robust codec that uses unique concealment techniques to minimize errors, is able to quickly recover from lost packets. Moreover, it boasts highly efficient jitter buffer management as well as channel-aware mode (CAM) for partial redundancy. Next, EVS features source-controlled variable bit-rate (VBR) adaptation for better speech quality at the same average active bit rate than fixed rate coding.
That allows mobile operators to optimize network capacity and voice call quality as desired for their service. However, the fact that EVS is able to offer unprecedented quality for speech, music and mixed content also means that it’s an intensive codec in terms of computational requirements. According to technology watchers, it’s six times more powerful than ARM-WB in terms of processing requirements.
Merits of a DSP Audio Solution
The EVS audio codec mandates a dedicated DSP solution designed specifically for voice processing. The super-wideband EVS codec, which provides excellent voice and audio quality on any mobile network, requires substantially higher signal processing power to run sophisticated multi-microphone noise reduction and echo cancellation algorithms.
CEVA, the supplier of DSP cores, is one of the firms proactively supporting the new voice codec. It has recently announced the availability of EVS voice codec for its TeakLite family of audio DSPs. CEVA provides EVS capability in the form of audio and voice software package that can run on the TeakLite DSPs.
TeakLite-4 processor is specially designed for codecs like EVS
A CPU that is not specifically designed for voice processing will simply consume too much MHz and power. Even a dedicated voice DSP like the TeakLite-4 that is specifically designed for such codecs takes quite a lot of MHz capacity. In fact, the performance of EVS and appropriate dual-microphone noise reduction can take up to 6mW when optimized for the TeakLite-4 DSP.
It’s worth noting that memory footprint of both code and data is quite large in EVS voice codec, and that mandates adequate memory mechanisms like caches. The CEVA-TeakLite-4 DSP core has all the required memory mechanisms like caches and advanced memory management.
Moreover, TeakLite-4 has a dedicated voice processing functionality and not just general DSP functionality like the M7 processor. In a nutshell, a DSP offers better performance for less area compared to the M7 microcontroller because of its dedicated voice processing ISA and memory management capabilities.