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Streaming Audio Codecs Explained

Streaming Audio Codecs Explained

If you have worked with audio for any length of time you know that not all audio formats are created equal. Here is what you need to know about the history of streaming audio codecs, and choosing a format that suits your needs. The Radio Mast Streaming Network is compatible with both MP3 and all AAC-type codecs, as well as some lossless formats.

  • 1993 - MP3

    MP3 is a compressed audio format that emerged in the mid-90s, and gained widespread usage as the format of choice for digital music. It was the first audio codec that offered good sound quality with file sizes that were convenient at the time. The quality achievable at standard bitrates (128 kbps) was also well below the download speeds of early broadband modems (1000 kbps), so the invention of the MP3 codec kickstarted the trend for internet radio and music downloading. Today, MP3 as still a streaming format that is universally supported across effectively every device, and it still offers relatively good sound quality due to improvements in encoders over the years.

  • 1997 - AAC

    AAC is the successor to MP3. AAC offers higher accuracy, higher efficiency, and more samplerates than MP3. The format was created in 1997, but did not achieve widespread usage until Apple introduced AAC support in iTunes and began distributing iTunes Music Store downloads in AAC format. It is nearly universally supported across every device and web browser, and is an excellent choice for streaming radio because of the better sound quality when compared to MP3s. (AAC is now sometimes referred to as Low Complexity AAC, or LC-AAC.)

  • 2000 - Ogg Vorbis

    Ogg Vorbis was created as a patent-free open source codec as a response to MP3. It consists of an audio compression format called "Vorbis" and a container format called "Ogg". Ogg Vorbis gained popularity in the early 2000s as it offered better sound quality than MP3 and WMA, but grew more slowly due to a lack of support in hardware players. Today, it is widely supported in players and browsers as a streaming audio format, but generally still lags behind AAC in terms of quality and compatibility. It was originally thought that the Ogg name was based on a character from Terry Pratchett’s Discworld novels, but the creators of the Ogg format say that isn’t the case. The origin of the Ogg name comes from the jargon of a video game called Netrek.

  • 2001 - Ogg FLAC

    FLAC was developed by the "Xiph.Org Foundation" as a lossless audio compression format for digital audio. In 2000 Josh Coalson began developing the FLAC codec and it entered beta stage with the release of version 0.5 of its reference implementation in January 2001. Since then the FLAC format has matured and grown to be widely used by music enthusiasts for its high fidelity. The FLAC codec creates a bit-perfect reproduction of your original audio for the listener, while still being more compact to store or transmit. Ogg is a container format which packages up the compressed FLAC audio in a format that makes it possible to stream over the internet. Ogg is what makes FLAC streaming possible, so all "FLAC" streams are really "Ogg FLAC" streams.

  • 2003 - HE-AAC v1

    "High-Efficiency AAC" (HE-AAC) is an extension to the AAC format that provides better sound quality at lower bitrates, which makes it more optimized for streaming. HE-AAC v1 uses a technique "spectral band replication", which discards high frequency audio before encoding, and then recreates those high frequencies during decoding using a psychoacoustic model. It essentially fakes high frequency audio in a way that your brain can't hear anyway, and by doing so, allows it to "spend" more bits in the encoding of lower frequencies that are more noticeable. The result is significantly better audio quality at lower bitrates (64 kbps and lower).

  • 2004 - HE-AAC v2 (AAC+)

    Also known as AAC+, HE-AAC v2 is another extension to AAC, which introduced a technique called "parametric stereo" to provide even higher quality at lower bitrates. Parametric stereo splits audio up into a mono signal and an ultra low-bitrate side band that is used to create a convincing approximation of the original stereo signal. On devices which support AAC but don't support AAC+, the audio will usually still play, but will be heard in mono. AAC+ should be used only at low bitrates (8 - 64 kbps). At bitrates above 64 kbps, regular AAC should be used.

  • 2012 - Ogg Opus

    Opus is the successor to the Vorbis and Speex codecs, and it offers very high quality and efficiency. It is perhaps the most versatile audio codec, and is used for low-latency voice (VOIP), streaming audio, music, site-to-site links, and more. The quality of Opus is considered equal to (or slightly better) than AAC and AAC+, especially at low bitrates. It is widely supported on mobile devices, but still lags slightly behind AAC in terms of compatibility.

Below is a table to summarize the relative quality of the codecs that we discussed in this article:

Rank by Quality Codec
#1 Ogg Opus
#2 HE-AAC v2 (AAC+) and AAC
#3 Ogg Vorbis
#4 MP3
#5 Ogg FLAC

Here we summarize the relative compatibility of the codecs discussed above:

Rank by Compatibility Codec
#1 MP3
#2 AAC
#3 HE-AAC v2
#4 Ogg Opus
#5 Ogg Vorbis
#6 Ogg FLAC

Still not sure which codec to use? Consider these tips for choosing a codec:

  • A 64 kbps AAC stream is roughly equivalent in quality to a 128 kbps MP3 stream.

  • Mobile listeners love saving bandwidth - Try offering a 32 kbps AAC+ or Opus stream.

  • AAC+ is only recommended for bitrates of 64 kbps and lower. Above that, regular AAC provides better quality.