This document explains two different methods to share audio in Moodle. They are referred to as Simple and Advanced as the latter requires slightly more preparation but allows for more flexibility. We would suggest following the Advanced Method for any audio you are creating for Moodle. But if the audio has already been created then the Simple Method will be fine in most cases. Instructions are also given for converting existing audio if it is felt necessary.
The following formats are widely used and if not already present on a computer, easy to use software can be easily downloaded and installed at no cost.
- .wma (Windows Media Audio)
- .ra (Real Media)
If you have a choice in the matter we would suggest choosing MP3. It is the most widely used format and is likely to be playable on any modern computer, internet connected device or portable player. If you are creating audio files, or having them created for you we suggest some specific settings for your MP3 in the next section.
Turning on multimedia filters
Moodle has a facility called Multimedia plugins that can make the presentation of some kinds of downloaded audio and video files more streamlined. Turning this on is recommended, and is a job for your Moodle administrator. It can however, in some rare cases cause your MP3 files to sound faster and higher pitched, making normal speech sound like Minnie Mouse, or slower and lower pitched. If this is the case then you may need to follow the Advanced Method outlined below to prepare your audio files for use within Moodle. Alternatively, you can try to ensure that the media filter doesn't attempt to process the problematic file.
The Media Filters in Moodle use Flash to playback audio in the browser. Flash requires the audio to be in a fairly specific format and this happens to also ensure the widest possible compatibility in other situations e.g. playing on an iPod. The short version follows, it is explained in greater depth below:
- MP3 format
- with a sample rate of 11.025, 22.05 or 44.1 kHz
- Constant Bit Rate (CBR) rather than Variable Bit Rate (VBR)
- Joint-Stereo, rather than Mono or Full Stereo
Note that 'sample rate' or frequency, measured in kiloHertz (kHz), is not the same as 'bit rate', measured in kilobits per second (kbps). The latter is a measure of filesize and download time and also a rough measure of quality. Somewhere between 32 and 96 kbps is appropriate with diminishing returns beyond 128kbps.
Sample rate is the number of times per second that the sound is digitally recorded. Due to something called the Nyquist Theorem you need twice the frequency of the sounds you wish to record. 44.1kHz is probably the most compatible and a good default choice but 22 and 11kHz are fairly standard too. You might sometimes find MP3 audio files at 16, 32 and 48 kHz but these can cause problems with Moodle and elsewhere and so ideally should be re-encoded following the instructions given below.
Variable Bit Rate files are commonly used to get the best audio quality by 'saving up' bit on easily encoded sections such as silence or simple audio and then using them for difficult to encode noises such as applause, harpsichords or hi-hats. Constant Bit Rate mp3s can be seen therefore as wasteful as they use the same amount of bits for silence as they do for a full orchestra crescendo but are more compatible and easier to stream. If audio files report that they are much longer or shorter than they really are then VBR incompatibility is a likely cause. If audio sound faster or slower than it should, an you have ruled out the sample rate as a cause then VBR may be the culprit.
Mono files theoretically save bandwidth when the location of the audio is irrelevant (e.g. a single person talking), but joint-stereo mp3s can encode most stereo info with minimal quality loss and are smart enough to deal with primarily mono audio without wasting bits and unnecessarily increasing filesize and download time. Full or Real stereo is only necessary for the very highest quality of recording where stereo separation is regarded as important. Mono files are another potential cause for audio to playback at twice the expected speed.
Pros and cons of the advanced method
- Ideal for shorter sound clips which can be flexibly embedded into a document, forums, quizzes, lessons etc.
- Allows longer audio files to play instantly in the browser as they download in the background.
- Widest possible compatibility (both software and hardware)
- consistent user experience across platforms (Mac, PC, Linux) as it plays in the browser
- Can require more preparation
- The Media Filters can have issues with some mp3 audio files produced by third parties, that would play fine if downloaded
Conversion to compatible MP3
If you have audio in one of the other formats listed above but would prefer it in the widely compatible MP3 format detailed above then most can be easily be converted. The following instructions are for iTunes which is freely and easily available for both Mac and PC.
To change the settings used to create or convert audio files, open the iTunes option screen, then select:
Options -> Advanced -> Importing, then select MP3 Encoder then from Settings: select Custom...
- Stereo Bit Rate: your choice. (see below for guidance)
- Don't tick "Use Variable Rate Encoding"
- Sample Rate: only choose either 44.100, 22.050, or 11.025 kHz (lower sample rate as you lower bitrate)
- Channels: Stereo
- Stereo Mode: Joint Stereo
- Don't Tick "Smart Encoding Adjustments" (this doesn't have any effect unless you have left some settings at 'auto')
- Tick "Filter frequencies Below 10 Hz" (this generally doesn't have any effect but doesn't really hurt and can increase quality)
If you have played the file in iTunes you should be able to find it in your library. After selecting the file you should find an option in the Advanced menu, called Convert Selection to MP3. If you have several files to convert you can select them all at the same time.
Choosing a bitrate
If you already have an existing file then you shouldn't change the bitrate without good reason. It is the equivalent of photocopying a photocopy and each copy introduces degradation and artefacts. If you're not going to reduce the bitrate by 50% or more you would probably be best leaving it as it is.
MP3 bitrates can range from 1 to 320kbps. You can experiment to find the right mix of size and quality appropriate for your content but anything over 128 kbps is probably a waste of time and space for anything other than recording high-quality music, as is anything higher than the bitrate you start with if converting from one format to another. From 32 to 96 kbps is roughly the right area for recording speech.