Audio Compression FAQ

What is Audio Compression?

Resi’s encoders use audio data compression, where an audio codec (for example AAC or MP3) reduces the overall size of an audio track. Compression makes it easier to stream over a variety of bandwidths. This is different from dynamic range compression which is used in audio mixing. While both are often employed in a video broadcasting environment, this article will focus on audio data compression.

What is an Audio Codec?

An audio codec compresses raw audio data to reduce its file size and allow for more reliable streaming at lower bandwidths more efficiently with little to no loss in quality.  This is done by removing imperceptible signal components and other redundancies created in the coding process. Sound variations that cannot be heard by the human ear are removed and the coded audio is cleaned up to remove redundant data.

Which Audio Codec Does Resi Use?

Our encoders use AAC-LC (Low-Complexity Advanced Audio Compression) to transcode audio for multisite and web destinations. AAC-LC is one of the most widely used audio compression coding standards. It is primarily used for distributing audio at a higher bitrate (64kbps or higher per channel). This differs from AAC-HE (High Efficiency Audio Compression), which is optimized for delivering audio at significantly lower bitrates and sample frequencies.

Does Social Media Transcode Audio After it Runs Through a Resi Encoder?

Resi’s cloud sends compressed audio to your social media destination(s), which is then transcoded to operate within each platform. Because of this, Resi has no control over audio issues that arise as a result of being broadcast to social destinations. To confirm if an audio issue is occurring within social media and not somewhere else in your signal path, check out our guide on troubleshooting audio issues. If you can’t find an answer, we recommend looking at YouTube’s help site or Facebook’s live video FAQ.

Is My Event’s Volume or Quality Changed When it is Compressed?

When audio is compressed with a codec, such as AAC, there should be no difference in the volume when it is broadcast to your viewers. Since compression primarily affects audio data that is imperceptible to the human ear, quality degradation is normally the result of something with the signal before or after it is encoded (e.g. during the mixing process or after it is transcoded by a social media site). 

One of the most common problems with audio volume and clipping comes from using analog recording equipment. If your setup includes equipment such as tube amplifiers or compressors, there may be issues with how you set your reference levels (the level setting on a mixer, or another piece of audio equipment that's used for for producing a desired signal) before they reach your encoder or after they have been transcoded to fit a social media or third-party web site’s streaming standards. Analog equipment defines levels in dBu (decibel unloaded). These levels, also known as RMS (root mean square) levels, are representative of the amount of voltage going through a piece of equipment of these levels can vary by the design and quality of the equipment. As such, they can be expressed in both positive and negative output changes, with 0 being a reference point for when distortion typically occurs.

On the other hand, most digital spaces, such as DAWs, digital mixers, or analog-to-digital converters read levels in dBFS (decibels relative to full scale), which are units of measurement based on the maximum possible output for your digital gear. Because digital audio equipment has a defined limit for output, unlike its analog counterpart, it bases its readings on a maximum peak level (represented by 0) and everything else is expressed in relation to that peak through negative signal output changes. When an analog signal is converted to digital

Since there is no standard conversion for analog to digital levels, In this sort of hybrid production setup, it is essential that you test your digital and audio levels before you broadcast or record. If you are mixing with analog equipment, you may find that the volume has decreased once it has been transcoded to the web. As such, you will need to do the following when testing how your analog equipment sounds after it has been transcoded:

1. Make sure your recording and encoding equipment is properly configured.
2. Create a new scheduled event for today (whenever you would like to run your test).
3. Add the content destination(s) you would like to use and make sure to restrict public access. (Learn more)
4. Make sure you can see and hear the event on a separate device or monitor so you can hear how it sounds.
5. On your digital mixer or analog to digital converter, use 
6. Keep a record of where these levels are for the next time you have to broadcast using a similar setup.

If you would like more information, you can read our articles on how to Troubleshoot Audio Issues and Signal Path Basics.

It is important to remember that Resi support cannot not resolve signal path issues. As such, any signal issues that occur outside of your encoder or decoder will need to be resolved by an integrator. You can contact one of our local resellers to inquire about signal integration services.