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This is to serve as a basic tutorial on audio formats and their related subtypes. In no way is this meant to be a definitive guide, and is merely a compilation of freely available information - and falls short of even the lowest standards for audiophiles. For a truly comprehensive (and far more technical guide/debate) please visit the forums of

Before we begin:
Lossless Data Compression is a class of data compression algorithms that allows the exact original data to be reconstructed from the compressed data. This can be contrasted to Lossy Data Compression, which does not allow the exact original data to be reconstructed from the compressed data.

MPEG-1 Audio Layer 3 (MP3), is a lossy digital audio encoding format. This encoding format is used to create an MP3 file, a way to store a single segment of audio, commonly a song, so that it can be organized or easily transferred between computers and other devices such as MP3 players.

MP3 uses a lossy compression algorithm that is designed to greatly reduce the amount of data required to represent the audio recording, yet still sound like a faithful reproduction of the original uncompressed audio to most listeners. An MP3 digital file created using the mid-range bitrate setting of 128 kbit/s results in a file that is typically about 1/10th the size of the digital data found on an audio CD.

What is this "LAME" thing I keep hearing about in reference to mp3's? - Today, LAME is considered the best MP3 encoder at mid-high bitrates and at VBR, mostly thanks to the dedicated work of its developers and the open source licensing model that allowed the project to tap into engineering resources from all around the world. Both quality and speed improvements are still happening, probably making LAME the only MP3 encoder still being actively developed.

1. CBR - In Constant BitRate encoding (CBR) the bitrate is kept constant across the entire file: the same number of bits is allocated to encode each second of audio, and internally, frames of audio data occur at regular, predictable intervals, so the overall file size for a given duration of audio is predictable. In theory CBR can be set to encode at any bitrate, though the most common on this site are 192kpbs (kilo bits per second), 256 kbps, and 320 kbps. The higher the bitrate, the higher the quality and file size.

2. VBR - Variable BitRate (VBR) encoding is designed for size & quality optimalization. Where there is silence in the music, it is less "demanding" in terms of its encodability, it makes sense to drop the bit rate, simply because there's not much there to encode, and the wasted space is overkill. Where the full orchestra and high noise percussion is joining in, the encoder will choose a higher bitrate appropriate to the demands. Some parts of the music can be encoded in 128 kbps without any quality loss, other parts get the full 320 kbps to make the best of it. But on the average, the size of the VBR-encoded mp3 file will be (much) lower than one with a constant 320 kbps encoding. Why waste disc space? Why go for less then highest quality? That's the goal of VBR - High quality, smaller file size.

Usage: -V(number) where number is 0-9, 0 being highest quality, 9 being the lowest. There are instances of V3 and higher in the audiobooks section, where bitrates are less imperative to enjoyability.

Common VBR types seen at (referencing the LAME 3.97 codec) :
V0 - Option designed for maximum quality and minimum size; the highest quality with the smallest file size. The bitrate usually averages around 256 kbps, but achieves CBR 320 kbps quality. Depending on the complexity of the sound wave, the resulting mp3 is about 224-256 kbps on the average. (V0 = alt-preset fast extreme)
V2 - This is still very high quality, with bitrates averaging around 192 kbps, but targeting CBR 256 kbps quality. The files are, obviously, smaller. This is also the format used by the "scene." (V2 = alt-preset standard)
APS - Alt-Preset-Standard, Old terminology replaced by V2
APX - Alt-Preset-Extreme, Old terminology replaced by V0

FYI : There is no such thing as a 320 VBR

3. ABR - Average BitRate (ABR) refers to a form of variable bitrate encoding where the encoder will try to reach a target average bitrate or file size while allowing the bitrate to vary between different parts of the audio or video. As it is a form of variable bitrate, this allows more complex portions of the material to use more bits and less complex areas to use fewer bits. However, bitrate will not vary as much as variable bitrate encoding. Two-pass encoding is usually needed for accurate ABR encoding, as on the first pass the encoder has no way of knowing what parts of the audio needs the most bitrate to be encoded.

Free Lossless Audio Codec (FLAC) is a file format for audio data compression. Being a lossless compression format, FLAC does not remove information from the audio stream, as lossy compression formats such as MP3, AAC, and Vorbis do. In turn, FLAC files are much larger than their mp3 counterparts.

Like other methods of compression, FLAC's main advantage is the reduction of bandwidth or storage requirements, but without sacrificing the integrity of the audio source. For example, a digital recording (such as a CD) encoded to FLAC can be decompressed into an identical copy of the audio data. Audio sources encoded to FLAC are typically reduced in size 40 to 50 percent.

FLAC is suitable for everyday playback and audio archival, with support for tagging, cover art and fast seeking. FLAC's free and open source royalty-free nature makes it well-supported by many software applications. FLAC playback support in portable audio devices and dedicated audio systems is limited but growing. Josh Coalson is the primary author of FLAC. The FLAC plug-in is available at their website, here

FLAC has a built in compression setting, ranging from (0-8). What the compression setting does is tell the compressor how hard to try to compress the file. Lower settings result in quicker rips at a higher bitrate. Larger settings result in slower rips at a lower (more compressed) bitrate. Regardless of which you choose, the resulting files are of the same lossless integrity as the file is only being compressed more or less, not lost. The default (5) is recommended on many forums, as it is a compromise between file size and speed, though this is a subject of debate and really a matter of subjective preference.

Vorbis is a free and open source, lossy audio codec project headed by the Xiph.Org Foundation and intended to serve as a replacement for MP3. It is most commonly used in conjunction with the Ogg container and is therefore called Ogg Vorbis. The Vorbis format has proven popular among supporters of Free software. They argue that its higher fidelity and completely free nature, unencumbered by patents, make it a well-suited replacement for patented and restricted formats like MP3.

For many applications, Vorbis has clear advantages over other lossy audio codecs in that it is patent-free and has free and open-source implementations and therefore is free to use, implement, or modify as one sees fit, yet produces smaller files than most other codecs at equivalent or higher quality.

Advanced Audio Coding (AAC) is a standardized, lossy compression and encoding scheme for digital audio.
AAC generally achieves better sound quality than MP3 at the same bitrate, particularly below 192 kbit/s.
AAC’s most famous use is as the default audio format of Apple's iPhone, iPod, iTunes, and the format used for all iTunes Store audio. Overall, the AAC format allows developers more flexibility to design codecs than MP3 does. This increased flexibility often leads to more concurrent encoding strategies and, as a result, to more efficient compression. However in terms of whether AAC is better than MP3, the advantages of AAC are not entirely conclusive, and the MP3 specification, while outdated, has proven surprisingly robust.

Also known as WMA Standard or WMA Std, it was created by Microsoft to compete against MP3, which was quickly becoming the de-facto standard format for lossy compression at the time. The primary distinguishing trait of the WMA Standard format is its unique use of 5 different block sizes, compared to MP3, AAC, and Ogg Vorbis which each restrict files to just two sizes.

Even though Microsoft claims it is able to deliver the same quality as MP3 at half the bitrates, that statement is certainly false. A more realistic number would be same quality at around 25 % smaller bitrates - and that applies to low bitrates only. At 128kbps, it is easily bested by LAME.

File format for audio data compression. Being a lossless compression format, Monkey Audio does not remove information from the audio stream, as lossy compression formats such as MP3, AAC, and Ogg-Vorbis do.

Like other methods of compression, the main advantage of using Monkey Audio lies in a reduction of bandwidth and/or a reduction in storage requirements, but, in the case of Monkey Audio, there is no sacrificing of the integrity of the audio source (as there would be with, for example, MP3). For example, a digital recording (such as a CD) encoded to Monkey Audio can be decompressed into an identical copy of the audio data. Audio sources encoded to Monkey Audio are typically reduced to about half of the original size.

Monkey Audio is suitable for distribution, playback and archival purposes. However, it is a proprietary software, it is often too slow to decode on portable audio devices, and it has limited/problematic support on software platforms other than Windows. There are alternatives that provide the user with more freedom and official support for more platforms, such as the FLAC format.
Lo-Fi Version Time is now: 16.06.2019 - 18:30