Skype has recently released a new codec to their system that improves the sound quality of audio transmissions. What does that mean to you? It means that now the audio quality that you can achieve via Skype is starting to rival the audio quality that can be sent via other much more expensive systems such as ISDN and Source Connect. The question: Is it really good enough for professional studio use or is it just close but no cigar?
I decided to put this new Skype version to the test. First, how do you know if you have the new Skype version? Well, from what I can tell, this updated audio codec called Opus is working in the latest updates (read comment below). I'm not aware of what exact version of Skype you have to have installed for this new codec to start working but on my Mac as of this writing I'm running version 188.8.131.522. The audio quality really is quite remarkable as well as the latency is very low. That means that you can have a conversation via Skype and almost never step on each other's words because you're not waiting for the other person's transmission to reach you, making it much easier to communicate in real time. Anybody that has ever worked by ISDN or Source Connect knows what I'm talking about.
So let's first talk about how Skype's new audio quality rivals Source Connect. For those who are not familiar with Source Connect, it is a professional audio software from Source Elements that allows you to stream studio quality audio via internet connection. It's been available for at least 5 years now and has pretty strong foothold in the voiceover industry. Of all of the competing systems that are becoming available, Source Connect definitely has the biggest head start. Source Connect is cross-platform so it will run on Mac or Windows and but is not available yet to run on iOS or Android phones or tablets.
Most of you are likely familiar with Skype as it allows audio transmission via the internet for free. It's very flexible because it allows video, screen sharing, calls to landlines, chat and to share files with the other user or users who you are connected with during the call. Skype does run on iPhone, iPad, Android, Windows, Mac and Linux among others. Both Skype and Source Connect can be used to communicate in real time time with very little latency with another studio and sound quality is quite remarkable.
However, when we start looking at the resulting recording made via Skype or Source Connect you start to notice the differences between the two systems. My initial tests show that while the audio quality from Skype is surprisingly good, especially considering the price, it does fall short in regards to pure audio quality. Some portion of the Skype codec, the way that Skype handles its audio, is adding dynamic compression to the signal. The resulting recorded audio file clearly has been limited or compressed and some way by Skype. This could be a problem depending on the needs of the studio who is receiving your file. If they want your audio to be completely in it's original state with no processing of any kind, Skype will not be an acceptable substitute for Source Connect.
I think you will agree that the quality would acceptable for radio spots, field reporting, and many projects where immediate access to remote talent is required. Skype will make an excellent backup to SC while traveling in areas with poor broadband Internet access or network firewall issues preventing a two way connection via SC.
Correction from Frank Frederick:
|After reading your treatise on the codec I downloaded the file. Here is a simple explanation: Skype is still using SILK, not Opus.
Why my comment? My contacts reveal that Opus will be released (if it is included_ in a the new version of Skype - sometime in 2014 - not before.
a) the example shown shows a max bandwidth of 15k (FM radio quality) which is SILK codec's limitations.
b) the audio presented is SILK at 96k. Flat tops and lack of solid low mid's or body in the low end on Spectral display, another trademark of SILK. SILK at 128 is much better than what you have posted.
c) the new Skype will not include Opus in the Android or Mac iOS versions.
The aac codec used, was under performing. At 96 bits it should have had a full bandwidth of 22K shown in spectral analysis. Although more solid in the low end and body, it lacked luster and shows a 16500 max high frequency in spectral analysis.
Please George, don't believe everything you hear or read on the internet. I would suggest you correct your postings as not only are they misleading, but they are incorrect as the codecs are NOT what you have lead people to believe.
If you want to hear what Opus can do, I will send you a true comparison file using the true Opus codec. Or you can wait and try a little something new which is coming next month to Indiegogo and then in production in July.
Thanks, Frank, I was not aware that Skype was able to improve the encoded audio bandwidth using SILK to such levels. I'll be sure to make corrections where needed and give you credit as appropriate.
That said, if it sounds good, it is good, and I think you must agree that the resulting audio would be "good enough" in some situations, and not in others.
This FAQ on Opus may shed more light:
This article is also helpful in explaining the distinguishing characteristics of SILK and Opus: