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What is a Good Bitrate Guideline for MP3 Files?

R. Kayne
R. Kayne

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 a high degree of 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.

An mp3 player.
An mp3 player.

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.

MP3s are compressed digital music files that sacrifice quality for file size.
MP3s are compressed digital music files that sacrifice quality for file size.

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.

Discussion Comments


I've got a question about recording music off the computer. I have a program that's really pretty cool by Wondershare and I'm a novice where it comes to the recording format. I have a choice between MP3 or M4A which I know my ipod uses. The other options I'm not sure of are Bit rate, I've got between 64-256kbps to choose from and the Sample Rate I've got 22050Hz-48000Hz to select here. The Defaults are MP3, 44100Hz, 128kbps. Will changing this make the sound any better? I have a great deal of cash tied up into sound card and Paradigm speakers looking for the best sound. Thanks for any help.


"It happened that I downloaded a very good song/music, but the mp3 has low bit rate (128)..."

No, the sound quality will be worse.

MP3 is a "lossy" format. Whenever audio is encoded to MP3, some of the audio information is discarded during the encoding process. The higher the compression settings the smaller the file size but the lower the sound quality - that's the trade-off.

Adobe Audition, like most audio editors, decodes MP3s before use. When you save / export the edited audio, the audio data is re-encoded and a little bit more sound quality is lost.

There are some MP3 editors such as "MP3Split" that are able to perform basic editing without decoding the MP3 first. This is a better way to edit MP3s if only basic editing (cut, delete, trim type actions) are required. If you need to do more complex editing or processing and want to retain the same audio quality you will need to export in a lossless format such as WAV or flac.


This is helpful. So can I use this in itunes and go under Get Info on a song to the info tab. Then under bpm place whatever value I want? or does itunes know that there is a predetermined bpm for a song?


If you are trying to create your own samples and you have higher quality samples that you are editing - if your sound card allows for it - try and record using at least double the 44,100Hz sample rate: 88.2kHz. 96kHz should be used when recording or editing for DVD in a home studio, such as in post-production in film. When down sampling, you must dither and use anti-aliasing using a steep low-pass filter at the Nyquist Frequency - 22,050Hz (half of 44.1, since CD is stereo) - to prevent distortion, and keep the audio quality as high as possible. Cheap converter will always cause loss of information.

Only ever use even multiples of the original and end frequencies when down and up sampling.

You should also try and do all edits and productions using .wav files whenever possible, as it is uncompressed audio, and can be converted to an mp3 with a better quality end result if editing is done properly. Remember to make all cuts on the zero crossing line on your waveforms.

There are some even more powerful sampling frequencies used in the high-end professional audio world. My sound card can do 192,000Hz - at amateur home studio prices; but the highest possible is 5,644,800 Hz currently used in the Super Audio.

Always use the highest bit rate your computer allows.

Though electronica is often bass driven, there are many different cymbals used among different songs, as well as many effects that cannot be produced in real life. Some may be of very high pitches (though this is not good to have all the time on a dance floor, as super loud treble will cause damage to your ears much more quickly than lower pitches) and should be represented as best as possible. The broad range of frequencies in the broad range of genres in modern music in general, should in itself require the best possible system, the best sampling rates, and the best bitrate you can afford with your processing speed.


I'm editing the audio on my laptop so that I can get the best sound from my headphones when I plug them in. I came across different bit sizes and sampling frequencies.

The highest bit size I have is 24. The highest sample frequency I have is 192,000 Hz. The second highest one is 96,000 Hz and then 48,000 Hz followed by 44,100 Hz. I've read that the sample frequency does not change what you hear much as you go higher than 44,100 Hz, however I was wondering what you all think.

I listen to mostly electronica and love bass. I don't know if the bit size and sampling frequency make the audio sound much different, but I want to know which one is optimal for how I listen to music. Thank you!


Once a higher quality mp3 is down-sampled, it loses detail because some of the information from the original file would be discarded (lost). A program cannot fill in the gaps with what had been discarded in the downsampling process/conversion. that missing information is simply not there anymore.

The up-converting program doesn't know what information was suppose to be there in a higher sample mp3, so it approximates. You'll get poor results when up-converting from a low sample rates.

Upsampling will be more successful with higher quality sample rates, only because there is more of the original information still present. But still, it won't be anywhere near a mirror of the original audio file. It will still be an algorithmic approximation.


A very helpful posting and very well written, explaining a complicated subject clearly and concisely. Thank you very much for taking the time to do so. It is much appreciated. Tony, UK


A song was MP3 bit rate 128kb. I converted it to MP3 192kb with 44100 Hz. When I play it, it doesn't sound as good as before. the volume is lower a little bit too. Why? which bit rate and Hz has the best quality ? I'm trying to record music so I want to save it to the best quality. Thanks


Thanks for this, but can someone please explain the significance of Hz in this?

For example, I have a voice based file (a lecture) I recorded using wavelab off an internet audio stream.

When making an mp3 from this wav file, for CBR I can specify both HZ (say 44.1KHz)and sampling rate, say 128 kbps, or many different combinations. 48KHz, 44.1 KHz, or 32KHz, each with 32 to 320 kbps. Is there a rule or some guidelines about which combination to use? Thanks for any help.


George, I'm no expert but I'm guessing it works on the same kind of principle as resizing pictures.

You can take a large high res photo and chop it down while preserving quality but if you take a small sized photo and try to blow it up then the quality is massively reduced and distorted.

To get 192 or 256 you'd need to get a file that's been compressed down to them bit rates from a higher bit rate file or download the high bitrate file and compress it down yourself.


Could you clarify this point please?

It happened that I downloaded a very good song/music, but the mp3 has low bit rate (128).

Now, if I edit it to 192 or 256 (using a prof. audio software like Adobe Audition or other) will the sound quality really become higher?

Or: NO, the real quality will stay 128, because the original file was 128?

Do mp3 files keep the maximum potential quality of 320 kbps, even when they are set to 128?

Thanks in advance, George


Great explanation. Thanks for this info.


Is there any quality difference in hearing the 44 kHz Mp3 and 48 kHz Mp3?


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 compatibility 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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    • MP3s are compressed digital music files that sacrifice quality for file size.
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      MP3s are compressed digital music files that sacrifice quality for file size.