MP3 File Structure

An MP3 file is made up of multiple MP3 frames which consist of the MP3 header and the MP3 data. This sequence of frames is called an Elementary stream. Frames are independent items: one can cut the frames from a file and an MP3 player would be able to play it. The MP3 data is the actual audio payload. The diagram shows that the MP3 header consists of a sync word which is used to identify the beginning of a valid frame. This is followed by a bit indicating that this is the MPEG standard and two bits that indicate that layer 3 is being used, hence MPEG-1 Audio Layer 3 or MP3. After this, the values will differ depending on the MP3 file. The range of values for each section of the header along with the specification of the header is defined by ISO-IEC 11172-3. Most MP3 files today contain ID3 metadata which precedes or follows the MP3 frames.

MP3 Bit Rate

The bit rate is variable for MP3 files. The general rule is that more information is included from the original sound file when a higher bit rate is used, and thus the higher the quality during playback. In the early days of MP3 encoding, a fixed bit rate was used for the entire file.

Bit rates available in MPEG-1 Layer 3 are 32, 40, 48, 56, 64, 80, 96, 112, 128, 160, 192, 224, 256 and 320 kbits, and the available sampling frequencies are 32, 44.1 and 48 kHz. 44.1 kHz is almost always used, and 128 kbits has become the de facto "good enough" standard, although 192 kbit/s is becoming increasingly popular over peer-to-peer file sharing networks, mostly due to the wider availability of broadband Internet access. MPEG-2 and the (unofficial) MPEG-2.5 include some additional bit rates: 8, 16, 24, 32, 40, 48, 56, 64, 80, 96, 112, 128, 144, 160 kbits; while providing lower sampling frequencies (8, 11.025, 12, 16, 22.05 and 24 kHz).

Variable bit rates (VBR) are also possible. Audio in MP3 files is divided into frames, each of which has its own bit rate, so it is possible to change the bit rate dynamically as the file is encoded. This technique makes it possible to use more bits for parts of the sound with higher dynamics (more sound movement) and fewer bits for parts with lower dynamics, further increasing quality and decreasing storage space. For example, a portion composed of pure tones could be encoded at 48 kbs, taking up less space without any noticeable difference, while a portion played by a full symphony orchestra is encoded at 224 kbs to express it with greater fidelity. Although not originally implemented, many encoders now use this technique to greater or lesser extent.

Non-standard bit rates up to 640 kbs can be achieved with the LAME encoder and the free-format option, but few MP3 players can play those files. Gabriel Bouvigne, a principal developer of the LAME project, offered the following information about free-format streams:

Free-format is compliant with the mp3 standard. Decoders are required to be able to decode it up to 320kbps, but decoding higher bit rate free-format streams is not mandatory. Practically, it means that higher than 320kbps, only a few decoders support it.

MP3 Audio Quality

Because MP3 is a lossy format, it is able to provide a number of different options for its "bit rate" - that is, the number of bits of encoded data that are used to represent each second of audio. Typically, rates chosen are between 128 and 320 kilobits per second. By contrast, uncompressed audio as stored on a compact disc has a bit rate of 1411.2 kbs.

MP3 files encoded with a lower bit rate will generally play back at a lower quality. With too low a bit rate, "compression artifacts" may be audible in the reproduction. A good demonstration of compression artifacts is provided by the sound of applause: it is hard to compress because of its randomness and sharp attacks. Therefore compression artifacts can be heard as ringing or pre-echo.

As well as the bit rate of the encoded file, the quality of MP3 files depends on the quality of the encoder and the difficulty of the signal being encoded. As the MP3 standard allows quite a bit of freedom with encoding algorithms, different encoders may feature quite different quality, even when targeting similar bit rates. As an example, in a public collective test featuring two different MP3 encoders at about 128kbps, one scored 3.66 on a 1-5 scale, while the other scored only 2.22.

Quality is heavily dependent on the choice of encoder and encoding parameters. While quality around 128kbps was somewhere between annoying and acceptable with older encoders, modern MP3 encoders can provide very good quality at those bit rates, not statistically different from quality provided by AAC, the technical successor of MP3. However, in 1998, MP3 at 128kbps was only providing quality equivalent to AAC-LC at 96 kbps and MP2 at 192 kbps.

The transparency threshold of MP3 can be estimated to be at about 128k with good encoders on typical music as evidenced by its strong performance in the above test, however some particularly difficult material can require 192k or higher. As with all lossy formats, some samples can not be encoded perfectly transparent to all users. An alternate scheme of encoding is VBR encoding. This aims for consistent quality, and varies the bit rate accordingly. Users who know a particular "quality setting" is transparent to their ears can use this value on all their music, and not need to worry about performing personal listening tests on each piece of music to determine the correct settings.

At lower bit rates, the quality of MP3 quickly degrades, and is far behind AAC quality at 32kbps, as demonstrated by a collective listening test. It is also important to note that perceived quality can be influenced by listening environment, listener attention, and listener training.