What is a Good Bitrate Guideline for MP3 Files?

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MP3 files are compressed audio files that are made from audio formats such as the wave (.wav) format. Wave files replicate analog recordings and digital sound files with 100% accuracy at the cost of large file sizes, while MP3 files sacrifice some quality for a smaller footprint. The quality sacrificed can be mitigated by several factors in the conversion process. With the right bitrate and configuration, MP3 files can provide extremely high-quality results that make them very close to their original wave files when played on portable audio players.

The balancing act between file size and quality is a somewhat subjective. To an audiophile any difference will be discernible. Others might not be able to tell a high quality MP3 file from its original wave source at all. In many cases the difference only becomes clear if played through a high-quality stereo system where the smallest nuances of the acoustic environment become clear.

MP3 files are primarily targeted for portable audio players. In this arena quality MP3 files come through with astounding sound given their small file size. Since portable players have limited memory, it makes sense that people want their MP3 files to be as small as possible while preserving as much quality as possible.

To this end the single most important factor in the creation of MP3 files is the bitrate. Generally, the more bits preserved per second from the original file, the higher the quality of the MP3 and the larger the file size. A lower bitrate reduces size and quality. The idea is to use a bitrate that results in maximum authenticity without preserving unnecessary data, which only creates larger files without appreciable difference to the ear.

For audio voice recordings such as lectures or language lessons preserved in wave form, bitrates of 32 kilobits per second (kbps) should be acceptable, though 64kbps might provide better quality depending on the source. Voices might sound "flat" at 32kbps, though they will be understandable. A 64kbps MP3 file made from a voice recording should sound nearly identical to the original.

Non-saturated acoustic music that features simple arrangements should get good results with a bitrate of 192kbps. If the music will be played on high quality equipment, you might opt for 256kbps. Music that falls in this category would include ballads, “boy-band” songs, easy listening and folk music. Also the work of many classic artists such as James Taylor, Linda Ronstadt, Joni Mitchell, and Simon & Garfunkel.

To make quality MP3 files from classical music and jazz, the best bitrate depends on the song’s characteristics. Soft jazz can normally be replicated at 192kbps to create a good balance between file size and diminishing returns, though 256kbps might sound better on the home entertainment center. Orchestral classical should do well at 256kbps for portable players, but files of 320kbps might be a better choice if you'll be burning to CD for the home or car.

For saturated music such as hard rock, metal, arena, pop, electronic and house music, 320kbps will give the best results. The greater number of bits per second will preserve more of the complex acoustic envelope.

When possible it is preferable that MP3 files be created using a variable bitrate. This allows the encoding program to determine if a particular frame of music requires the full bitrate. If not, the program reduces data retention for that frame resulting in a smaller file without sacrificing quality. Forcing a program to "over-sample" a frame can produce artifacts.

While this article is intended as a general guideline, one might find that he or she is just as happy with lower bitrates for specific songs or in general. Many factors affect our ability to judge the quality of music, including not just the equipment we use, but our activity when listening. For those who listen to MP3 files when exercising or walking outside, for example, exterior noise will make it more difficult to pick out qualitative differences. Conversely, audiophiles might prefer to sample everything at 320kbps, regardless of their equipment, the music's genre, or listening habits.

If making your own MP3 files, there are also other settings that affect quality. LAME is an excellent MP3 encoder and is free, along with the many graphical interfaces that serve as a front-end for this well-known command line program. LAME allows the user to tweak many settings in order to produce high quality MP3 files in seconds. One can also try various bitrates on a source file to find the best subjective balance between quality and file size.

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New: Discuss this Article

Posted by: anon16747
On sampling rates v. bitrates...

Analog audio, e.g. sound, is a waveform. Waves of air pressure, or captured by a microphone to make an electrical signal representing the waveform.

Digital audio is an approximation made by measuring - sampling - the waveform at regular intervals. So the 'Sampling Rate' is how often the wave is measured.

Music reproduction equipment attempt to handle a range of audio frequencies from 20hz to 20,000 hz. To represent a sound you need a sampling rate at least double the frequency. Compact disc sampling is 44,100 Hz, Digital Audio Tape sampling is 48,000 Hz and are thus capable of representing 20,000 Hz signals.

A musical note is tuned on its basic frequency - its fundamental frequency. But each note has higher frequency harmonics or overtones typically multiples of the fundamental frequency.

A concert piano has notes ranging from 27 to 4,000 Hz. Human voice ranges from 80 to 1,000 Hz. But the fidelity of the sound includes capturing many of those higher harmonic frequencies as well. So you want to capture frequencies 4 times higher than the fundamental up to about 16,000 Hz which is the limit of the ears of most adults.

A telephone has a sampling rate of 8,000 Hz and thus handles sound frequencies up to 4,000 Hz. FM radio handles sound frequencies up to 15,000 Hz.

For a speech, a sampling rate of 8,000, 11,025 or 12,000 Hz should be fine. Further, the sound can be recorded as one channel Mono as opposed to Stereo for further space savings.

For singing, you want a higher sampling rate like 16,000, 22,050 or 24,000.

For instrumental music, or movie soundtracks you want at least 32,000 Hz sampling or the 44,100 or 48,000 standards.

Bit-rate, a number like 128Kb, is a different concept altogether.

A CD stores music uncompressed at a sampling rate of 44,100 Hz; a sample size of 16 bits and in stereo - 2 channels. The bitrate is 44,100 X 16 X 2 = 1,411,200 bits per second.

MP3 encoders vary in quality, but generally can produce a fair-quality representation of CD music using 128kb/s (11:1 compression ratio) and a very good representation at 320kb/s (4.4:1 compression ratio). AAC compression can produce equivalent quality with only 3/4 of the bits, so a 96kb (14.7:1) AAC file with the quality of a 128kb MP3.

The MPEG standards specify a list of bitrates, and all players should be able to handle all of them. Some MP3 encoders will allow non-standard bitrates that can make smaller files with adequate quality, but for compatability your bitrates should be on this list: 8, 16, 24, 32, 40, 48, 56, 64, 80, 96, 112, 128, 144, 160, 192, 224, 256, 320 kb/s

Standard sampling rates are: 8,000; 11,025; 12,000; 16,000; 22,050; 24,000; 32,000; 44,100; 48,000 samples per second (Hz)

Here are some typical sampling rates, their uncompressed sizes and standard mp3 bitrates that correspond to fair and very good compression ratios:

8,000 Hz, Mono = 128kb uncompressed : 8kb(16:1) to 24kb(5:1)

8,000 Hz, Stereo = 256kb 16kb(16:1) to 48kb(5:1)

22,050 Hz, Stereo = 705kb 48kb(15:1) to 144kb(5:1)

32,000 Hz, Stereo = 1mb 64kb(16:1) to 192kb(5:1)

44,100 Hz, Stereo = 1.4mb 80kb(17:1) to 320kb(4.4:1)

48,000 Hz, Stereo = 1.5mb 96kb(16:1) to 320kb(4.8:1)


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