Hello Gibby,

My knowledge of the old NicaX codecs is very limited I'm afraid - maybe someone else with more knowledge can fill in the blanks on the true capabilties of your box. Let me give you some general info on the different compression algos in any case...

Apt-x is a proprietary compression algorithm developed by APT. G722 is an open compression algortihm that is/was mostly used by sports commentators and news reporters. It is not suitable for voiceover use. With "open" I mean that the implementation details can be downloaded from the ITU website so anybody can implement it in their codecs, giving it a high compatibillity with codecs from other manufacturers.

What your "L2" mode stands for I cannot be sure... It may mean that it is an Apt-x mode that uses two B channels to give you a higher audio quality. It may also mean that it is a MPEG Layer-2 modus. If it is the latter then you are in luck as MPEG Layer-2 is suitable for VO work - actually, it is considered the standard compression algorithm for voiceover use and is now very compatible amongst the different brands of codec.

All major brands of codec (Telos, Musicam, AudioTX) support G722 as well as MPEG layer 2 and MPEG layer 3. Non of these brands support Apt-X I'm afraid. If your codec indeed has MPEG Layer 2 then you can connect to most (if not all) studios and do your sessions with them. If your codec is only G722 and Apt-x capable then you will only be able to work with studios that have an Apt-x codec since the G722 mode will not give enough quality for VO work.

Hope this explains things a bit!

Edwin.

For voiceover sessions mono 128kbps MPEG Layer 2 at 48kHz is usually preferred. The 128kbps mode will use both B-channels of your ISDN line. Using both B-channels significantly boosts the audio quality of the session, which is why this mode is used for voice over sessions.

The 24kHz mode is normally only used by reporters and sports commentators on a single B-channel (64kbps) connection. Using a single channel will half the cost of the connection; very useful for connections that can last for hours! Using a lower samplerate (24 instead of 48kHz) decreases the amount of artifacts on the 64kbps connection but will also reduce the audio bandwidth. It is a compromise and is as such not used much for voice over sessions.