Skype switches to new Opus codec
Yesterday, in the official blog, it was announced that the company would soon be going to switch Skype to use a new efficient audio codec oriented to work in wireless networks and intended to improve the quality of the transmitted sound.
The development of Opus was launched in 2009 and already in September 2010 the codec was sent for certification to the IETF (Internet Engineering Task Force). The main technical advantage of Opus lies in the found balance between audio signal compression and its quality, which is important in terms of transmission in the networks of mobile operators. The codec uses a flexible adaptation algorithm in the event of a change in bandwidth - for example, when switching from a 3G signal to a Wi-Fi connection - and, in the future, it can provide a conversation in CD quality. Special algorithms have also been applied to combat packet loss with limited capabilities of the wireless network.
The attractiveness of the codec is also provided 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.
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You can listen to the audio recording encoded by Opus on the official codec website .
UPD: The ValdikSS user suggests the Opus specifications and schedules comparing it with other codecs:
Opus quality:
Latency:
[ Source ]
The development of Opus was launched in 2009 and already in September 2010 the codec was sent for certification to the IETF (Internet Engineering Task Force). The main technical advantage of Opus lies in the found balance between audio signal compression and its quality, which is important in terms of transmission in the networks of mobile operators. The codec uses a flexible adaptation algorithm in the event of a change in bandwidth - for example, when switching from a 3G signal to a Wi-Fi connection - and, in the future, it can provide a conversation in CD quality. Special algorithms have also been applied to combat packet loss with limited capabilities of the wireless network.
The attractiveness of the codec is also provided 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.
')
Hidden text
You can listen to the audio recording encoded by Opus on the official codec website .
UPD: The ValdikSS user suggests the Opus specifications and schedules comparing it with other codecs:
- Bitrate from 6 to 510 kbps (in fact, from 8 kbps)
- Sampling frequency from 8 to 48 kHz
- Frame size from 2.5 ms to 60 ms
- Supports both CBR and VBR
Opus quality:
Latency:
[ Source ]
Source: https://habr.com/ru/post/151414/
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