At RINGR, we love talking about sound technology so this week’s blog post is about sample rate and codec. If you have a question about podcasting technology, please drop us a note and we’ll do our best to answer it in a future blog post. You can ask your question by sending an email to email@example.com.
What is a Sample-rate?
The sample-rate of audio relates to the number of times a seconds that the audio signal was sampled. I.e. in 22.05 kHz audio, a number is recorded 22,050 times every second representing the audio signal from the microphone. Those numbers can represent frequencies in the audio up to half the sample rate (i.e. 11.025 kHz).
The range of human hearing is generally accepted to be 30 Hz – 19 kHz, however that range narrows significantly as we age and we are most sensitive to the 2 – 5kHz range, which is the most important range for understanding speech. A POTS recording is limited to frequencies of 3.3kHz (with some of the bass removed), while VoIP codecs can generally handle 4 or (at most) 8 Khz.
Unless you are recording music, there’s no real need for 44.1 kHz audio, 22.05 kHz is enough for an excellent quality speech recording with much better vocal quality than the audio from POTS or VoIP allowing you to capture all frequencies up to 11.025 kHz.
What are Codecs?
A “codec” is a device or program that compresses data to enable faster transmission and decompresses received data. Video and music files are typically large so they become difficult to transfer quickly over the Internet. Using a codec helps speed up downloads by encoding a signal for transmission and then decoding it for viewing or editing.
RINGR avoids the heavy compression applied to VoIP calls – however we compress the final output files, as raw WAV files for long calls are very large and hard to move around, and the files produced preserve much higher quality than the VoIP compression.
RINGR Premium provides a number of options for output file types, compressed using 3 different codecs:
- supported almost everywhere.
- introduces some audio artefacts at lower bit-rates,
- can sound a little ‘tinny’ at times.
- sounds loads better than MP3 for the same files size,
- open source/free, supported by most editing software,
- better supported than FLAC.
- not quite as widely supported as MP3 (but works on almost all phones and browsers.)
- lossless (identical to the WAV file but less than half the size),
- open source/free,
- supported by most audio editing software – an ideal transfer format.
- not supported by most browsers and requires a plugin to playback,
- larger file sizes than MP3/OGG.
You should also consider the Compression level or Bit-rate: The higher the bit-rate the better the quality. However, if the user is on a low bandwidth connection then a higher bit-rate may mean delays or buffering periods, negatively affecting their listening experience. Mobile users may also chew through their data allowance quicker.
Experiment with different bit-rates and quality settings and find the lowest setting that still sounds great – your listeners will thank you for it! Speech is very compressible (compared to polyphonic music) so expect to use lower bit-rates than you would for music – also if it’s a mono recording you can reduce bit-rate again compared to a stereo recording.
Hope you enjoyed this bit of podcasting tech talk! Your next step is to try RINGR for free. Create a free account today and discover the crystal-clear clarity for your podcast interviews.