Skype switches to the new Opus codec

Yesterday in the official blog it was announced that the company is going to transfer Skype in the near future to use a new effective audio codec oriented to work in wireless networks and designed to improve the quality of transmitted sound.

The development of Opus was started in 2009 and already in September 2010 the codec was sent for certification in IETF (Internet Engineering Task Force). The main technical advantage of Opus lies in the balance found between compression of the audio signal and its quality, which is relevant in conditions of transmission in the networks of mobile operators. The codec uses a flexible adaptation algorithm in the event of a change in channel bandwidth - for example, when switching from a 3G signal to a Wi-Fi connection - and, in the future, can provide a conversation in CD quality. Special algorithms are also used to combat packet loss with limited wireless capabilities.

The attractiveness of the codec is also ensured by the fact that it is free and can be freely licensed by third-party developers for use in VoIP applications.

Serious technical details are given in a long 45-minute video with the presentation of the new codec by one of the Skype engineers.



You can listen to how an Opus encoded audio recording sounds on the codec's official website .

UPD: ValdikSS user prompts the technical characteristics of Opus and gives a graph of its comparison with other codecs:

• Bit rate from 6 to 510 kbit / s (in fact, from 8 kbit / s)
• Sampling Rate 8 to 48 kHz
• Frame size 2.5 ms to 60 ms
• Supports both CBR and VBR

Opus Quality:



Latency:



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