If video is included in the content, the data rate should be as low as possible.įor these reasons, in March 2005 3GPPTM introduced (in Release 6) two new 'audio' codecs for Packet Switched Streaming Service (PSS), Multimedia Messaging Service (MMS), Multimedia Broadcast and Multicast Service (MBMS), IMS Messaging Service and Presence Service. Streamed audio content should be made available at a low bit-rate well below 32 kbit/s, corresponding to the bit-rate range already used in the AMR-WB codec. Radio resources and channel capacity set further limitations on data rates available for streaming. For this expected mixed streaming content, the codecs described so far have difficulties in performing consistently well for both speech and music at low bit-rates (i.e. Mobile streaming audio and messaging services may contain speech only, music only, or speech mixed with music on background. Enhanced interoperation with AMR-WB is provided over all nine bit rates between 6.6 kbps and 23.85 kbps. Furthermore, it features a special channel-aware mode achieving increased robustness in particularly adverse channel conditions. The codec also contains a system for jitter buffer management (JBM). An error concealment mechanism mitigates the quality impact of channel errors resulting in lost packets. The codec features discontinuous transmission (DTX) with algorithms for voice/sound activity detection (VAD) and comfort noise generation (CNG). The ability to switch the bit rate at every 20-ms frame allows the codec to easily adapt to changes in channel capacity.
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It further provides a significant step in quality over these traditional telephony bandwidths with Super Wide Band (SWB) and Full Band (FB) operation starting from 9.6 and 16.4 kbps, respectively. The EVS codec enhances coding efficiency and quality for NarrowBand (NB) and Wide Band (WB) for a large bit rate range, starting from 5.9 kbps variable bit rate (VBR). The EVS codec addresses 3GPP's needs for cutting-edge technology enabling operation of 3GPPTM mobile communication systems in the most competitive means in terms of communication quality and efficiency. The codec for Enhanced Voice Services (EVS), the successor of the current mobile high definition (HD) voice codec AMR-WB, was standardized by the 3rd Generation Partnership Project (3GPPTM) in September 2014. Our Role & Activities Enhanced Voice Services (EVS) codec Specification of speech and audio codecs has been one of ETSI's (and 3GPP’s) many success stories.
The optimal design principle of a codec thus, lies in meticulous bargaining of the aforesaid trade-off which aims to maintain the QoS requirements.
However, data compression is dependent on space-time complexity tradeoff (like degree of compression, the amount of distortion introduced while using lossy data compression, computational resources required to compress and decompress the data). The lossless data compression reduces bits by identifying and eliminating statistical redundancy. Source coding is associated with the use of variable-length codes for reducing symbols to bare miminum so as to represent the information in the transmitting message. In any communication transceiver, a codec (COder/DECoder) is one of the basic building blocks for efficient transmission of data.
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