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Entries in Source Connect (11)


The new Skype: The latest Source Connect rival?

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 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.

Take a listen to an audio sample I recorded of voice actor Graeme Spicer with SC then Skype, or download the WAV file for further analysis. 


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.

I also spoke about this topic during the first segment of EWABS Episode 91, which you can watch here




VO STUDIO: A show focusing on VO's home studios

I got to consult on Kami and Kim's episodes, but I appear in Kami's episode.  

If you like the video, please Subscribe on Youtube and Share on Facebook!



IP Codecs: Main or Backup/Alternate? by Dave Immer

If an IP codec is your main live audio networking tool, then it’s probably because
A. You can’t get ISDN or are unwilling to pay for it,
B. Most of your clients use IP anyway.
But if ISDN is your primary codec, you should definitely also have an IP codec. This gives you security as a backup, plus flexibility to connect with compatible IP-only systems. And everybody can get IP.

With ISDN MPEG codecs from companies such as Telos, Musicam, APT, Mayah, Prodys, TieLine, AudioTX, etc, compatibility is not (usually) an issue, as manufacturers long ago agreed on connection standards. But with IP codecs, it’s still kind of like the wild west with Source-Connect on it’s proprietary (albeit popular) mountain, and most other IP codec companies providing limited cross-compatibility despite employing standard algorithms such as AAC and MPEG layers II & III.

No matter what IP codec you use, the usual internet problems will persist such as latency and uncertain reliability. But these disadvantages can be minimized with the right algorithms on the right hardware platform over the right network. My recommendation for IP codecs:

1. Algorithm: APTX or AAC Low Delay
2. Hardware Platform: Musicam Suprima family or Source-Connect on a fast Intel-based processor.
3. Network: 15Mb/5Mb cable/fios/u-verse

Let me know if you have another IP codec you particularly like.                    Thanks,

Dave                             Complete library of newsletters:


Loop-back Connections - Essential Tests for ISDN and Source-Connect by Dave Immer


Being able to confirm your codec system is operating properly or to identify the source of a problem is a basic procedure we should all have in our back pockets.  A network loop-back test can accomplish this easily and provide valuable insight:

1.) It shows you that your signal path is set up correctly. You should hear the signal that you are feeding to the codec input “slap back” to your codec output.
2.) It confirms your network is passing the bit-stream in both directions.
3.) For ISDN users it confirms your long distance carrier is cooperating.
4.) For Source-Connect  it confirms your bit-rate and receive buffer are set at usable values.

Source-Connect users can connect to one of the “echo” sites on the contacts list.

ISDN users can dial an AT&T ISDN number set up for this purpose: 732-758-9999. This number can be dialed multiple times depending on your codec bit-rate. For instance if your codec is in L2Mono128 mode you would make 2 calls to this same number. If you need to test an APTX codec at 384kbs you would make 6 calls. If the calls don’t go through (meaning you either have no long distance carrier or your long distance carrier is failing to complete the calls) try dialing from one line to another. These would be local calls.

Doing this test prior to your first session of the day is a quick, easy way to confirm all is well.

Let me know if you have questions about loop-back test connections.   Thanks,

-Dave                                Complete library of newsletters:


Out of Hear ISDN to Source Connect Bridging Services

In 2007, George Whittam built "Out Of Hear" for Steve Nafshun, renowned voiceover engineer with DG Entertainment in Studio City, CA. Steve came to George with an idea to provide affordable ISDN bridging to the voiceover community. George designed a system that allows to provide ISDN bridges to up to four users simultaneously, connecting through Source Connect or AudioTX Communicator.  

Steve's unique service is available anytime and has helped voiceover actors break the chains of their home studio by allowing them to travel the world and still meet the needs of their ISDN clients. Out Of Hear also provides rental kits built by ERS dubbed "VO2GO", which contain everything one needs to travel and connect remotely to any ISDN studio.