Digigram has added support for the Opus encoding algorithm for its Iqoya *Call and Iqoya *Call/LE IP audio codecs. A native digital audio compression algorithm with very low latency, Opus addresses issues associated with the use of IP networks for real-time audio transport, explains Digigram.
Digigram’s Iqoya *Call audio-over-IP solution brings professional quality to full-duplex connections between a remote site and a studio, and the Iqoya *Call/LE offers broadcasters a lower-cost model with the same quality, says the firm. Both offer complete support for encoding algorithms. In addition to Opus, the IP audio codecs support Fraunhofer AAC-LC, AAC-ELD, AAC-LD, HE-AAC formats, MPEG-L2 and L3, enhanced-aptX, G722, and G711.
According to Digigram, the Opus algorithm handles both speech and music in a single multiple-bitrate algorithm that natively manages error concealment and adapts to issues, such as packet loss and variable bandwidth. Audio quality is assured by the algorithm’s high speech-encoding quality at bit rates ranging from 24 to 32 kbps, and the low latency of Opus enables it to support audio contributions from reporters in the field, it says. At higher bit rates, the music-encoding quality of Opus makes it comparable to the better versions of AAC encoders.
Both the Iqoya *Call and Iqoya *Call/LE are now shipping with the free Opus software option.
