Audio over IP is nothing more than audio that is being sent over the internet. The digital audio file is encapsulated in a packet and it’s data is sent over the web just like a picture or an email. The end user is usually using a similar system which decodes the packet and plays back the audio in near real time. You might use this technology every day using FaceTime or Skype.

Audio over IP networks (AOIP) are nothing new but we are finally starting to see some major improvements with these distribution mediums. The technology replaced aging ISDN networks which were used many years prior by studios and radio stations. The infrastructure usually consisted of multiple telephone lines to create a high bandwidth network. The cost for hardware and service was atrocious. Both systems are capable of providing high quality audio with very little latency. Depending on the codec and the system being used it possible to get near original quality of a singer in Boston played back in real time to an engineer in Mumbai. Think of a very high qaulity phone call suitable for recording or broadcasting.

AOIP and VOIP technologies are steadily growing more reliable so it’s only a matter of time before manufactures start pushing these technologies as the standard to consumer markets. The major factors of audio over IP is quality and latency. Keep in mind that AOIP when used in live events, broadcast, and studio is far different than listening to your favorite artist over Spotify. Live engineers depend on near zero latency and high bandwidth which means buffering is not an option. Even when used in video conferencing it is incredibly annoying when the call is dropping in and out or there is a huge delay.

We’re still a long ways away from determining one standard that we will see as default in hardware including mixers, cameras, phones, conferencing systems and studios. The industry is still currenly dominated by a proprietary market where hardware manufactures usually require their technology be used on both ends of the transaction.

There was once a time when we really needed to pay attention to disc space and file size when it came to recording. Writing those log sheets is a thing of the past with the plethora of HD space now available at a relatively cheap price. However, I routinely look back on the correlation of file size and audio quality for various reasons. Usually when I begin a recording project I like to dedicate an entire hard drive to the band or project. Using the below  calculations I can generally determine the size of the hard drive needed to fit the project. Nowadays I tend to find myself always buying 2TB hard drives which will fulfill any project. If I waste the space, it doesn’t matter. Either way, these calculation tables are great for any engineer to have on hand.

All the calculations for high resolution audio below are for the Pulse Code Modulation audio format. Regularly referred to as PCM, it  commonly uses the file extensions .wav or .cda. There are quite a few other combinations of bits per sample and samples per second which may be used as well as a large number of containers and file extensions. Tables below refer to calculations based on the most common mono and stereo settings used by engineers. Calculations for MP3 files are based on the LAME audio encoding standards. They also include both left and right stereo channels because who the hell makes Mono MP3 files? But the again who records at an 64 Kbps? Kbps means “Kilobits per second” (1,000 bits per second) and KB means KiloBytes (1,000 Bytes). There are 8 bits in a byte. Note the uppercase “B” for bytes in “KB”. A lowercase “b” (Kb) would indicate bits.

The exact file size below will vary from system to system likely due to differences between binary and decimal counting systems (multiples of 2 versus multiples of 10). Check out my Base-2 Methodology post to get a better explanation of this confusing phenomenon. The file header information and ID3 tags (for MP3s) will have an effect on the numbers. Album cover art tends to affect these numbers the greatest as images can commonly be 2MB or larger. These numbers should be used as a “ballpark” amount.

Mono WAV (single channel)

Stereo WAV (dual channel)

Stereo MP3 (dual channel)

Let me explain a little bit about who I am and what Audiosyntax is all about. Audiosyntax Productions is a comprehensive audiovisual communications company in Indianapolis, Indiana. We specialize in onsite multimedia services and logistics for event and stage productions. Established in 2005, Audiosyntax Productions LLC has played a part in the success of many small businesses, traveling artists, stage shows and corporate events in the Midwest.

Chief audio engineer and technician Nate Sparks has an elaborate background of audio and video system technologies and has over 10 years of professional experience in the studio and on the stage as an audio mix engineer and producer. Nate takes deep pride in the service he provides and will be sure to make your next project a success.

Audiosyntax Productions offers a variety of services all related to audio/video technologies for home and business. We offer multi-track recording services, studio production and engineering, live sound system installations and operation, as well as technology consultations and repairs. Audiosytax Productions LLC also offers a large range of IT related services for support of the latest IP/AV integrated technologies including CobraNet and Dante.

Check out our SERVICES page to see a larger list of the services we offer. Occasionally we will offer services that are not listed on the website so do not hesitate to contact us if your need is not listed. Many of our services are customized based on the details of the project. Each customer is different so we will be glad to talk you through your project and find a solution that works. You can contact us by using the form provided on our CONTACT page.