Sound Design Stack Exchange is a question and answer site for sound engineers, producers, editors, and enthusiasts. It's 100% free, no registration required.

Sign up
Here's how it works:
  1. Anybody can ask a question
  2. Anybody can answer
  3. The best answers are voted up and rise to the top

I have friends that are very picky when it comes to mp3 bit-rate, and will always look for the 320 kbit/s version of a file. However, I have never noticed any differences, they sound the same to me. I remember reading somewhere that the human ear is simply incapable of sensing the difference, even if present.

Are there any audible differences between 192 and 320 kbit/s .mp3 files?

Are there any good quality tests that have been done in this area?

What types of factors would need to come into play to tease out the differences (if any) in audio quality between a 192 vs 320 kbit/s .mp3?

This is a question that's currently being discussed on skeptics.stackexchange. It was thought reasonable to get some input from the audio experts. So with the author's permission I'm posting it here.

share|improve this question

migrated from Jan 27 '14 at 15:07

This question came from our site for engineers, producers, editors, and enthusiasts spanning the fields of video, and media creation.

Can't say I'm the expert on this, but if you want to hear differences between different MP3 bitrates, the decay of a crash cymbal is worth listening to. It can sound very 'slushy' at lower bitrates. – Mark Heath May 15 '11 at 19:32
@Mark - Cheers for the info about the crash cymbal. I've read somewhere (I think) that the (high pitched) trumpet is also a giveaway for some people. But I could have just made that up, I honestly can't remember. I do all my CD or .flac --> VBR .mp3 using --preset fast extreme with lame. I can't tell the difference between the .mp3 and the source file through KRK VXT6s. I'll definitely keep an ear out for the cymbal though. – boehj May 15 '11 at 19:44
Muntoo posted this link on the same question on skeptics, and I just had to share again: ([Audiophiles] – JYelton May 16 '11 at 21:14
@JYelton - It's one of his best. :) – boehj May 17 '11 at 4:13
If you pick a bit of music, convert it to both different formats, convert them back to lossless high quality files again, stick em side by side in a decent DAW, phase flip one of them and mix them together, the phase reversal effect may mean that you you can hear just the difference between the two files. I've not tried it myself mind, so I don't know how – eviltobz Jun 14 '11 at 15:15

11 Answers 11

up vote 29 down vote accepted

If you're after audible differences the only test you can use is a double blind test. Empirical tests such as waveform analysis will most definitely show differences in the output, but that doesn't mean the differences are audible.

In this case you could have some one listen to samples of music and have them guess the bitrate of the samples. The ABX type test seems to be popular for bitrate comparisons: you listen to sample A and then B and then X and decide which of A or B sample X is from. So A might be 192 kbps and B might be 320 kbps and the listener has to decide whether X is from the 192 or 320 kbps encoded track. This kind of test is a little better than an "assign the bitrate to a sample" type test because the listener can't guess their way through it with any accuracy and skew the results.

There are other double blind approaches, see this article for details.

There's been a few magazine attempts at uncovering codec and sample rate differences. This Maximum PC article gets a lot of attention on the net, though they're doing 160 vs 192 and 320 vs uncompressed.

I couldn't find any academic papers on the matter (but I didn't look that hard), which surprised me a bit. I would have thought there'd been a bunch of grad student research in psycho-acoustics available.

share|improve this answer
+1 for double blind test, but bear in mind that you are also testing your speakers/headphones at the same time - if they have a limited frequency response they might mask the benefits of the higher bitrate encoding. – Mark Heath May 15 '11 at 19:37
Cheers for this Ian. I'm sure the people over at skeptics will be pleased to here about this. I, too, was surprised that there wasn't much jumping out at me on the net. So now I guess the key to settle it for the skeptics folks is to find a good ABX trial that's been done. – boehj May 15 '11 at 19:38
@boehj: I lost my IEEE DB access last year when I left my old company. If you can find someone with access to their papers I'm sure you'll be able to find IEEE-sponsored research papers on this topic. This is right up their alley. – Ian C. May 15 '11 at 20:35
There are some ABX software programs that will blind you from the switches. I find these to be very helpful when trying to evaluate this. – GreenKiwi Jun 25 '12 at 22:08
Jeff Attwood recently did a blind listening test on his blog. Read about it here. – JMFR Jul 23 '12 at 20:54

I find it comes down entirely to the type of music and the speakers you listen to it through:

I can happily listen to pop music on an ipod through earphones at low bitrate (well, I say happily - not really a fan of pop music, but what I mean is it sounds as it should)

If you play the same mp3 through a decent or high end audio system it will sound worse, but again, with pop music it is still acceptable.

The real problem I have is in trying to listen to either heavy metal or classical music at low bitrate through a high end system. The jangles, splashes and other digital artifacting ruins the pleasure I get out of a good piece of music. This happens mostly at the top end (cymbals, distorted guitars etc) and especially when there is significant bottom end at the same time (1)

That said, I never bother with 320 constant bit rate because it just leads to large files. You should be better off with variable bit rate as it does intelligently reduce the file size where it isn't needed.

1- recording one of our new tracks found a major problem when generating mp3s - there are sections where not only are there a lot of high pitched distorted guitars, but some major subsonics that dive from around 80 to 15 Hz. That required top quality VBR to cope:-)

share|improve this answer

I work with sound for a living.

I have placed the same cut from 128, 192, 256 and 320 into a human ear sound analysis test range with a $100,000 sound sampling oscilloscope and the full wave form across the entire spectrum of the human ear is filled.

Further analysis shows that the main problem with mp3 sound samples in not in the bit but the distorted recording setting at any bit. Other words, people have a tendency to rip music at a height of input much higher then is needed causing envelop folding of all wave forms adding distortion in the applied recording.

Higher then 95% input is damaging to the recording as it leaves little overhead. My rule of thumb is to not record any audio over 89.2% as this will give your recording a level for overhead. All sounds have peeks and valleys which has to be given space for or its distorted by envelop folding for highs and crashes (base muffling) for lows.

To acquire a good recording you must stay within a channel of tones and then it is only replicated by the quality of the listening equipment and the receivers ear. Both of which by the way are never the same or ever perfect.

What I am saying is there is a heck of a lot more to the quality of sound then just the bit it is replicated at. Everyone is tone deaf at some level. You can play one sound clip to 100 different people and everyone of them will hear the same clip differently at some point. This has been proven in sound lab testes over and over. Then you put in the factor of human acceptance (like this sound or not) the best you can come up with from any recording is 88% acceptance.

So, you see, your fighting a loosing battle to try to explain the best quality. Just don't over peak your recording and let the individual receiver adjust the listening level for their own ear and equipment. Do your part by just sampling at a modest level. That's all that you or anyone else can or ever do to appease anyone.

Quality of input recording is the best answer as bit replication depth has little effect as long as it is kept within acceptable reason. Just my opinion after 30+ years experience.

share|improve this answer
I agree completely. In a lot of cases its the level that encoding takes at that is the issue. I used to allow .3db of headroom when exporting to MP3. I now go for .6db simply because I started to notice some weird artefacts on things like reverb tails etc. The thing with hearing these issue with MP3 is you don't usually hear them until you do. But once you have you start hearing them everywhere. For the average listener on ear buds the difference is probably still there to be heard but most won't hear it because that aren't listening for it. – Simon Rigby Feb 20 '14 at 21:25
I don't really understand this. You're converting digital to digital if you're ripping from a CD. Are you saying that the distortion from the mp3 encoding, plus the peaks (times 90%) of the original wav is enough to go over 0dB? That sounds crazy to me. Have you tried comparing the original wav with the decoded mp3 in audacity or similar? – naught101 Dec 25 '15 at 10:10
@naught101 the process isn't lossless though, converting to a lossy format involves a pass through a psychoacoustic noise model involving band allocation, masking and rounding so you'll never get whatever you put in. Inter Sample Peaks can also be a real problem for some codecs. With 16 bit recording, you never need to go to full scale - it's a bad habit. However, the debate about whether you leave input audio as-is and deal with the artifacts - or whether you normalise to -1 dB and pbysically alter the source to preserve more of the source signal - is out of scope for this question! – Christopher Woods Dec 28 '15 at 16:25

The ultimate answer for this is: It depends on what material you are encoding.

And the strongest scientific evidence is in the coders themselves. When encoding an mp3 using VBR (Variable Bit Rate), some encoders show how many frames where encoded using which bit rate. Here's a screenshot from LAME:

The output of VBR mp3 encoding in LAME

You will notice that only 10 frames out of 10735 were encoded using 320kbps, which is the highest bitrate mp3 supports. Had the quality been set to highest (which is not in this example) this would practically mean no audible differences should be heard. Had no frames been encoded using 320kbps, no audible differences should be discernible. Sadly, the output doesn't mention whether the 320kbps encoding was sufficient, or whether a higher bit rate would be needed.

In my experience, only dense and harmonically-rich masters put load on the 320kbps. Most vocal tracks, or bass recordings would rarely require more than 192kbps.

From various experiments that have been conducted, including some I have been part of, you should expect to find that our hearing is not aligned with the output of these coders. In other words, with everyday type of material and environments, people cannot tell the difference between 192kbps and 256kbps, let alone 320kbps; while many experiments are not comprehensive (testing environments and statistical analysis), there is strong indication that our perception is much worst than assumed by mp3 encoders. Bear in mind that most digital radio stations use 96kbps for transmission.

However, a final verdict should be made on a case-by-case basis using rigorous research, which is extremely hard to conduct. Alternatively, the process above should give you cheap and fast indication to whether it should even be possible for people to tell the difference.

share|improve this answer
+1 for this. Anyone who really knows what they are doing will realize that the loss of detail that is noticeable is in the fine sounds that can't be well covered when the bit rate isn't high enough. If there aren't many fine sounds or the sound is muddy to start with, then high levels of compression work fine. When there are lots of little soft details that are still cleanly mixed, the required amount of data goes up. You don't identify compressed audio from the overall sound, you identify it from the details. – AJ Henderson Mar 20 '14 at 13:47

To an untrained ear it would be borderline impossible to tell, and even a trained ear needs to pay attention. As such, it is hard to quantify perceptible differences as everyone will perceive it differently, if at all.

My approach to audio engineering, which I picked up from my time at SAE, is to go by feel. You can monitor everything, tweak to the nth degree, but as audio is inherently analogue, that is all you should really measure. As Ian C says, an ABX test is a great way to see for your self.

To answer your question from my experience the answer would be yes absolutely.

The degree to which you can perceive the differences will largely depend on the content of the song. Orchestral music is in my experience the perfect subject matter to test with. It is both incredibly complex in it's sound makeup (usually covering the entire frequency of human hearing) and has a very large dynamic range.

Bit depth also has an effect, my experience is that an larger bit depth 24bit as opposed to the cd standard of 16bit, also makes a great deal of difference. The next natural step up is vinyl.

As noted by some other people, the gear you use will also influence the perception. On my little white earbuds the difference is not nearly as instantly noticeable as on my large studio monitors. So if you were to test, you should use the highest quality gear possible to take harmonic distortion from the gear completely out of the equation. You could use a boom box from 1980 and not notice a difference between A or B.

share|improve this answer
A follow up if you were going to do a test. Listen to the high frequencies, especially violins and cymbals. Also listen for complex sounds like a bright crunchy guitar. The most common artifact you will hear will be like a phaser or comb filter. Though, you would need to ensure such artifacts are not being caused by the gear used in testing, including the digital analogue convertor and so on. – Owen Kelly May 16 '11 at 15:35

There are some neat studies in the field of psychoacoustics that claim that lossy audio compression tasks our brains more than less lossy formats. I cannot for the love of me remember where I got this but I believe it was from I did do some background checking on those claims after the fact and they seemed to be fairly accurate.

Most of us can certainly tell the difference between 128 and 192 bit rates but the difference is less audible between 192 and 320. However, for the trained ear I purpose that there's a difference. I certainly feel that listening to some 320 bit rates compared to their 192 counter parts is less tasking. But it's really hard to apply this in the general sense. A lousy MP3 can definitely mess with my hearing even at higher bit rates but I cannot apply such a theory to the general population.

Listener experience (and training) definitely factors in. Unless your are running a streaming service and need to be conservative with the bandwidth, I say go with the higher, 320 bit rates.

share|improve this answer
This answer really needs refefences. – naught101 Dec 25 '15 at 9:52
@naught101 I can't find the original source anymore, sorry. But you should be able to independently be able to verify this quite easily. Just download some MP3 encoders and fiddle with various quality settings. If you listen to music at a low bit-rate for an extended period and then switch to the same music with (but which a much higher bit rate) I'll be surprised if you didn't sense some kind of relief. Though at what point the compression is an issue or not that's probably very individual. – John Leidegren Jan 1 at 14:00
@naught101 Also, our brain can certainly reconstruct a damaged audio signal but at some point it will require more of your attention to still be able to determine what it is you're listening to. As your attention is drawn to the noise, it's your brain basically devoting more energy to the task. – John Leidegren Jan 1 at 14:07
Sorry John, but that's totally unscientific. There's no way to guarantee that you're not just triggering confirmation bias. Of course a "better" sound file will sound "better" if you already know it's better. A/B blind testing would work, but it's not that easy for anyone to set up at home. I have serious doubts that the brain has the ability to decide that MP3 algorithm-related artefacts are "noise" and not just part of the music. I mean, Vinyl is lossy and lower quality than a decent MP3, but people still like the sound. Why couldn't the same be true of low quality MP3? – naught101 Jan 4 at 2:04
@naught101 if this peaks your interest I suggest you go and do the experiments. How else will you learn? I don't have the answer, I can only draw from my limited experience. I found one video that I think you should watch. It's about mastering/headroom the "loudness war" it's very interesting. this is one of my original references that I couldn't find by people who are active researchers in the field. – John Leidegren Jan 5 at 17:41

Another thing to consider is who/how is/are they encoding it. I find a lot of the time that people are encoding and playing with settings they may not understand. This question is so finite though that it almost parallels the "analogue vs digital audio" debate.

Can it be perceived? Yes.

Will it be bothersome to listen to a properly encoded 192kb/sec mp3? Not likely...

Will it be bothersome to listen to an improperly encoded 320kb/sec mp3? Yes.

Is it personal preference? Mostly.

I can't listen to 128kb/sec mp3's... can't do it. Yuck! 192 is fine for me. Saying that, are there folks out there who have stereo systems that replicate the audio well enough that developing a preference for 320 over 192 is believable? Definitely... I don't have that stereo system though ={

share|improve this answer

See and Both are simple, but valid studies that indicate that we can't really tell any difference between files encoded above 160kbps.

A couple of caveats:

  • The above figures are for stereo sound. For files with more channels (e.g. surround sound), you want correspondingly higher bit rates. So basically, you could hear the difference 160 and 320 kbps if there are more than 2 channels, as bitrate per channel will be lower.
  • Bitrate is affected by bit-depth. A 16-bit sound file will be fine for listening, but if you're going to be mixing it with other audio, you might need the extra headroom afforded by 20 or 24 bit sound, and bitrate (and file size) will be correspondingly higher. But you won't be able to hear this difference by listening to the audio file on its own.
share|improve this answer

I am a very picky listener. I notice that with acoustic instruments, you will hear the difference on a good audio set compared with mp3 players that are not high end audio.

I like dance organs from Decap Antwerpen (Belgium) and they are equipped with jazzflute ,vibraton, and flute. They have 8 registers that are natural pipes and the rest is electronics. It uses an inline synthesizer for sax, baritone, and trumpet.

I have all these music cd's stored as .wav and that sounds perfect. I tried .mp3 at different bitrates and is sounds less good with less ambiance... There is something missing in the warmth of the music.

share|improve this answer
I'd love to see good (non-anecdotal) evidence of people being able to distinguish between a good quality, 320kbps .mp3 and a good quality .wav file. – Robert Jan 3 '13 at 16:29
You might be waiting for a while Robert. – naught101 Feb 19 '14 at 23:51

Leaving the math out and measuring software off the table, there is some kind of distortion or oscillation in the middle upper frequencies. You hear it more with sustained Strings or Keyboard tone,and especially with symbols. All MP3's seem to have this issue no matter the resolution. It's less pronounced with a 320 kbps but still there. There is also a smaller stereo image in an open speaker situation. With headphones, I can't hear the change. A Wav/FLAC will be noticeably improved to an MP3 even at 320, with only a couple of exceptions being the type of music played, but even then, in an A/b test it will be noticed. How many people can, is another story. I was taught as an Engineer to remember only 5 percent of the population can hear what you hear, so don't spend all of your time making it perfect. Just worry about good.

share|improve this answer

I am new to this but thought it would be helpful to share my 2 cents worth. I noticed the depth of sound for a given music file when I stumbled upon an old version coded in 128kbps and listened to the same file back to back with a version coded in 320kbps. The annoyance is that once I sensed this difference, the perfectionist in me is now consigned to updating the remainder of my almost exclusively music library. Used reference quality headphones to notice the difference.

share|improve this answer
It's pretty well known that 128kbit is low enough to cause problems with the audio. That's why the question specifically asks about 192kbit. In my reading, 160kbit may be differentiable from 320 in some cases, but 192kbit isn't. – naught101 Dec 25 '15 at 7:56

Your Answer


By posting your answer, you agree to the privacy policy and terms of service.