Blog

The new Skype: The latest Source Connect rival?

2013-04-22T23:48:57Z

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 6.3.0.602. 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

2013-04-10T19:39:48Z

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!

Howard Parker Voiceover Studio 2011- Design and Construction

2013-02-16T01:43:54Z

Slideshow of the complete demolition, construction, acoustics, and equipment installation for Howard Parker's home voiceover studio. Designed by George Whittam, construction by James Michael Nail.

SOLD Booth for sale: Portable 4x5 voice-over booth by Drum Perfec

2013-02-06T18:06:19Z

For sale by Alejandria Kete

A booth used for professional and home recording studios as vocal, instrumental or voice over booth

This is a good value for something made this well that looks attractive enough for the home, a great alternative to the WhisperRoom or Vocalbooth products if you don't have noise issues coming through the floor below, as this is a floorless unit.

- Manufactured by Drum Perfect

- Only three months old (no serial number or model number

- Exterior footprint is 4' x 4.5' and 6' 6" high. Can fit two VO inside but a little cramped. Single musician or VO actor better

- Interior is 4' x 4'

- Wall thickness is about 3" and my apartment is next to the Los Angeles 170 freeway where it's LOUD, and this booth has been an

amazing sound barrier for me. This would be a solution for a musician or voice over artist in a multi-housing (apartment/condo) situation

- Two pass-thrus for cables: one on top of unit and another on bottom

- Features two double-paned acrylic windows (*this was an added cost)

- Over 112 square feet of absorbing surface

- Easy to apply corner and seam covers for maximum isolation

- Custom mocha exterior is easy on the eyes (*this was an added cost)

- Weight around 500 pounds (approximate)

- It has no floor

- I have the paperwork for the setup instructions

Here are the complete specifications from the manufacturer

$1,500 FIRM

BUYER is to come pick up and disassemble

Pickup address: North Hollywood, CA PLEASE CONTACT US EXACT ADDRESS INFORMATION

*there is ONE flight of stairs that goes straight up no winding corners

CONTACT US to arrange logistics.

In Or Outside The Box: Is it the Codec or the Network? by Dave Immer

2013-01-08T23:04:19Z

I have noticed in some blogs a tendency to intermingle codec problems with network quirks.

For instance, Source-Connect is an excellent piece of software that delivers very good audio using the AAC algorithm. When the network over which it is running is up to the task, results are satisfying. But a user expecting to get an “ISDN experience” from it when their internet service is experiencing packet collisions, jitter and bursty or slow performance, may have a tendency to associate such behavior with the software codec itself when that is not the case.

Another example would be the Telos Zephyr. Being as the Zephyr is immensely popular and has emerged as the standard ISDN codec (in the US,) people tend to co-mingle it’s behavior in their minds with the ISDN line to which it is connected. While the Zephyr is a well designed, user friendly box, it can be unstable compared to other major brands. The characteristics of such a “standard” codec might be perceived as the nature of ISDN itself, which, again, is not necessarily so.

Certainly troubleshooting is inevitable with any unit. And unless you have the option of substituting alternate equipment, networks or software, it’s way harder to arrive at a clear understanding of the problem(s.) So:
1. Rent one or ask a colleague if you can borrow their codec and try it on your line.
2. Take your codec over to a colleague’s network and try it on their line.
3. Involve a third party like the Digifon Bridge to make test connections.
This way you can at least determine if the issue is in or outside the box (or both!)

Let me know if you have comments about this. Thanks,

Dave Complete library of newsletters: www.digifon.com/blog.html

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

2013-01-04T19:37:31Z

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: www.digifon.com/blog.html

WAKE UP! (Your ISDN Codec, That Is...) by Dave Immer

2012-11-16T03:53:11Z

Both you and the remote facility are configured identical, the calls connect, but only one side is locked (framed). But when audio signal gets sent to the un-locked end, it becomes locked. If you own an older ISDN codec model such as the Telos Zephyr (both Classic & Xstream) or the Musicam Prima (both CDQ & LT) you may, from time to time, need to receive audio to ‘wake up the box.’

Of course, the user that needs to send the audio signal to wake up the other end doesn’t know the other end is not locked, since the locked (framed) condition only pertains to the decoder (receive.) So it’s good to send some signal as soon as the dial-up connection is made to make sure the remote side codec is ‘awake.’

I have also seen phantom signal appear on codec meters looking like a steady noise level, only to go away once an ISDN connection is established.

Often the equivalent of jiggling the wires or toggling the switch clears up the issue.

Sometimes a reboot is needed to get stuff to work. (The display on the Zephyr may show ‘Ready/Ready’ but the box cannot make or receive calls.) I put this in the same category as wire jiggling since no settings get changed to remedy the condition.

Let me know if you have questions about waking up the box. Thanks,

-Dave Complete library of newsletters: www.digifon.com/blog.html

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

2012-11-15T05:40:17Z

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: www.digifon.com/blog.html

SOLD! Vocalbooth.com Diamond Series Platinum booth for sale in Los Angeles

2012-10-05T07:57:36Z

I have a client in Los Angeles with a Vocalbooth.com 3 Carat Diamond Platinum Series booth for sale. This is the top of the line booth built by Vocalbooth, and it's spacious. The booth is designed to sit in a corner of your room and takes up 6' of wall space on each side. Ventilation and acoustic foam is included. It is in good condition and must be picked up. We will assist with disassembly and moving out of the building to your provided truck. We can move and re-assemble for an additional fee.

Model dia-0013 3 Carat Diamond Platinum Series RETAIL PRICE $7195.00

Asking $3700, but offers will be considered. Cash only paid directly to owner.

Please contact George for any questions.

Can You Sling the ISDN/IP Lingo? by Dave Immer

2012-10-02T01:54:32Z

Some acronyms and contractions have embedded themselves in our daily discussions of configuring, using and troubleshooting our live audio networks.

Here are some common ones in no particular order:

PSTN: Public Switched Telephone Network
ISDN: Integrated Services Digital Network
SPID: Service Profile IDentifier
CODEC: COder/DECoder
Telco: Telephone Company
TA: Terminal Adapter
IMUX: Inverse Multiplexer
kbps or kbs: Kilobits Per Second
BERT: Bit-Error Rate Test
LEC: Local Exchange Carrier
CIC: Carrier Identification Code
PIC: Primary Interexchange Carrier
CPE: Customer Premise Equipment
IP: Internet Protocol
QoS: Quality of Service
FEC: Forward Error Correction
SIP: Session Initiation Protocol

Let me know if there are other ones you encounter often. Thanks,

-Dave Complete library of newsletters: www.digifon.com/blog.html