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Skype To Give Away New SILK Audio Codec, Is it better than Speex?
kwanbis
post Mar 3 2009, 19:56
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QUOTE
Skype’s new state of the art speech codec SILK will be made available to third party licensees for free, the company is announcing later today. Skype GM Jonathan Christensen will be speaking about the new program at the eComm event in San Francisco later today.

SILK has been highly regarded by the guys that follow this sort of thing and is included in the most recent version of Skype for Windows (the Mac version with SILK will be coming in April). If both sides of the call have a version of Skype that includes the new codec, the call quality increases dramatically.

Skype is now making the codec available for third party use on a royalty free basis. There are a number of speech codecs available on the market today, including iSac and AMRWideband, and an open source codec called Speex. Skype claims that SILK outperforms all of these.

http://www.techcrunch.com/2009/03/03/skype...lk-audio-codec/

Has anyone tried to ABX Speex vs SILS?

This post has been edited by kwanbis: Mar 3 2009, 21:56


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IgorC
post Mar 4 2009, 08:50
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https://developer.skype.com/silk?action=Att...LKDataSheet.pdf

Suspiciously datasheet hasn't any details around settings and conditions of test. What version of Speex, CBR or VBR etc...?
BTW where to get binaries of last speex 1.2rc1?

I haven't any information about SILP but it should be more realistic to do some really independent tests. As example, Skype uses on2 VP7 videocodec which claims to outperform standard H.264 (which of course it doesn't )

This post has been edited by IgorC: Mar 4 2009, 08:56
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NullC
post Apr 13 2009, 20:55
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QUOTE (IgorC @ Mar 3 2009, 23:50) *
https://developer.skype.com/silk?action=Att...LKDataSheet.pdf
Suspiciously datasheet hasn't any details around settings and conditions of test. What version of Speex, CBR or VBR etc...?


It's worth mentioning that speex can't even encode at the bitrates given in those results. The results would make more sense if they used the nearest speex bitrate less than the displayed value, which is a valid methodology for hard bitrate constrained channels though thats not what VoIP usually is… but it's not one I'd use without making it very clear in my test results.


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