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Adding hard numbers to the answer to prove my point and to hopefully put discussion to a rest
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Mecki
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As a closing word, let me point out a rather sad fact. There arekeep in fact a bunch of peoplemind that not only people with hearing capabilities above the average can always tell MP3 from uncompressed apart, no matter what encoder, no matter what encoder settings, no matter what source material. And they claim, that's because of their extraordinary ears. The sad truth is: The opposite is true. Their ears aresometimes not extraordinary, they are in fact broken. See,be fooled by the psychoaccoustic model was developed, the same holds true for people with a normal workingbelow the average, as the model simply fails when your acuesthesia (= sense of hearing). Of course, some people hear better, others hear worse but the model was designed for people with "healthy ears". If your ears are damaged because you listened is too very loud music as a teenager or if you were born with a hearing deficit or if your hearing sense was attacked by some illness, the model will simply not fit your ears. Thefar way these people hear music is different tofrom the restone of us, similar like color blind people see the world in a different way than those with standard visiontest persons used to create that model. And the later one is far more common case as the hard numbers prove:

Nearly 16% of adults in the U.S. report hearing trouble.

[...]

  • About 40 million US adults aged 20-69 years have noise-induced hearing loss.
  • More than 1 in 2 US adults with hearing damage from noise do not have noisy jobs, meaning the exposure is likely recreational.
  • About 1 in 4 US adults who report excellent to good hearing already have hearing damage.

[...]

  • 15% of school-age children (6-19) have some degree of hearing loss.
  • An estimated 12.5% of children and adolescents aged 6–19 years have suffered permanent damage to their hearing from excessive exposure to noise, according to the CDC.

This may not be a problem in daily life and these people can enjoySo when entertainers who are regularly exposed to very loud stage music just alike but as they may not hear certain frequencies, the model does not model their senseclaim that MP3 (or any form of hearing at allcompressed music) sounds like crap and in that case the compression based on that model will failadvocate for them every time. E.g. frequencies that we cannot hearuncompressed digital music, take it with a grain of salt, as other loud frequencies nearby coverthis is probably just because it sounds like crap to them aren't covered for someone who cannot hear that other nearby frequency at all. So this person wouldThe fact that they can hear the frequency that we don't accordingcompression artifacts while the model and if that frequencymajority of people cannot is dropped, the person will notice while we won'tno proof for excellent hearing. SoAnd if someone can always hearyou cannot tell the difference, thisit doesn't mean your hearing is bad, it only means nothing unless his ears got professionally tested and nothat your hearing defect has been detected. A lotmatches those of musiciansthe persons used to create the psychoaccoustic model, who were randomly chose but, of course, all tested in advance to not have any serious hearing defects because.

So in the end it's also important to ask yourself one question: Who are you encoding the audio for? Just for yourself? Just for your family and friends? For a very large audience? Or for the vast majority of too loud stage music causing permanent damageconsumers? 192 kbit/s will be fine for most people and that's why so many musicians claimapparently also for yourself, otherwise you had not asked that MP3 is crap and sounds horriblequestion. WellIf you encode for a large audience or for commercial purposes, it does sound horrible but only to their earsyou may prefer a higher bitrate, though, as this will even make some people happy whose acuesthesia is above or below the average.

As a closing word, let me point out a rather sad fact. There are in fact a bunch of people that can always tell MP3 from uncompressed apart, no matter what encoder, no matter what encoder settings, no matter what source material. And they claim, that's because of their extraordinary ears. The sad truth is: The opposite is true. Their ears are not extraordinary, they are in fact broken. See, the psychoaccoustic model was developed for people with a normal working acuesthesia (= sense of hearing). Of course, some people hear better, others hear worse but the model was designed for people with "healthy ears". If your ears are damaged because you listened too very loud music as a teenager or if you were born with a hearing deficit or if your hearing sense was attacked by some illness, the model will simply not fit your ears. The way these people hear music is different to the rest of us, similar like color blind people see the world in a different way than those with standard vision.

This may not be a problem in daily life and these people can enjoy music just alike but as they may not hear certain frequencies, the model does not model their sense of hearing at all and in that case the compression based on that model will fail for them every time. E.g. frequencies that we cannot hear as other loud frequencies nearby cover them aren't covered for someone who cannot hear that other nearby frequency at all. So this person would hear the frequency that we don't according the model and if that frequency is dropped, the person will notice while we won't. So if someone can always hear the difference, this means nothing unless his ears got professionally tested and no hearing defect has been detected. A lot of musicians have defects because of too loud stage music causing permanent damage and that's why so many musicians claim that MP3 is crap and sounds horrible. Well, it does sound horrible but only to their ears.

As a closing word, keep in mind that not only people with hearing capabilities above the average can sometimes not be fooled by the psychoaccoustic model, the same holds true for people below the average, as the model simply fails when your acuesthesia (= sense of hearing) is too far way from the one of the test persons used to create that model. And the later one is far more common case as the hard numbers prove:

Nearly 16% of adults in the U.S. report hearing trouble.

[...]

  • About 40 million US adults aged 20-69 years have noise-induced hearing loss.
  • More than 1 in 2 US adults with hearing damage from noise do not have noisy jobs, meaning the exposure is likely recreational.
  • About 1 in 4 US adults who report excellent to good hearing already have hearing damage.

[...]

  • 15% of school-age children (6-19) have some degree of hearing loss.
  • An estimated 12.5% of children and adolescents aged 6–19 years have suffered permanent damage to their hearing from excessive exposure to noise, according to the CDC.

So when entertainers who are regularly exposed to very loud stage music claim that MP3 (or any form of compressed music) sounds like crap and advocate for uncompressed digital music, take it with a grain of salt, as this is probably just because it sounds like crap to them. The fact that they can hear the compression artifacts while the majority of people cannot is no proof for excellent hearing. And if you cannot tell the difference, it doesn't mean your hearing is bad, it only means that your hearing matches those of the persons used to create the psychoaccoustic model, who were randomly chose but, of course, all tested in advance to not have any serious hearing defects.

So in the end it's also important to ask yourself one question: Who are you encoding the audio for? Just for yourself? Just for your family and friends? For a very large audience? Or for the vast majority of consumers? 192 kbit/s will be fine for most people and apparently also for yourself, otherwise you had not asked that question. If you encode for a large audience or for commercial purposes, you may prefer a higher bitrate, though, as this will even make some people happy whose acuesthesia is above or below the average.

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Mecki
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To get an idea for the answer you can use a VBR encoder like LAME and use it as best quality (-V 0). At that level LAME tries its best to preserve every signal it considers hearable by human ears according to its model - it aims to be perfectly transparent to all listeners for all source materials. So whenever a frame is encoded with less than 320 kbit/s, then no more than the bitrate of that frame was required to preserve all these signals. Here's a sample output of running such a command:

Note however that the example above was LAME trying to be as losslesstransparent as possible without just always using 320 kbit/s for file size reasons. MostUp to today, nobody has proven with an ABX test, that he can tell the difference between 320 kbit/s CBR and LAME -V 0 VBR MP3; which does not mean that people couldn't tell the difference between MP3 and uncompressed! It only means that when -V 0 was not transparent, CBR 320 kbit/s was not transparent either.

Yet most people neither have ears good enough, nor equipment good enough to require such a level of quality. For the vast majority of people and home equipment, -V 2 should already be perfectly transparent. So let me repeat that test again with thatthis option:

And now you can see that to reach that level of quality, 192 kbit/s is in fact enough most of the time, often even 160 kbit/s is. There are only a few cases where LAME hathad do go up to 224 and only very little where it had to go all the way up to 320 kbit/s. So if I had limited encoding to 192 kbit/s, that would still have produced perfectly transparent frames to most listeners most of the time, just not to all listeners all of the time.

One of the differencedifferences between -V 0 and -V 2 is that the first one considers all frequencies as important whereas the later one only frequencies below 18671 Hz, which is one of the reason why it has to encode less signals. Yes, human beings can hear utup to 20 kHz, but usually only small children. Already most teenagers cannot hear frequencies that high anymore and rarely any grown up adults, no matter how good their ears, no matter how good their equipment, so cutting these very high frequencies usually makes no audible difference for almost all listeners, unless these are young children and they will hardly care if these frequencies are missing in whatever source they are listening to. So yes, something is definitely missing here in the audio signal, regardless how many bits were used, but people who say they can tell first need to prove that by an ABX test and if such a test is made correctly, they usually cannot.

As for the psychoaccoustic model, one of the reason why AAC can achieve much better quality at the same bitrate or the same quality at a lower bitrate is because it uses a better and more fine grained model and encodes signals in a more fine grained manner, too, as one without the other one would make little sense. It would be possible to make a better model for MP3 but the format lacks the capability to make use of it. And it would be possible to design a model that works better even for trained ears but the result would only be encoders that produce bigger MP3 files in VBR mode with no benefit to the vast majority of users as they wouldn't hear any difference.

As a closing word, let me point out a rather sad fact. There are in fact a bunch of people that can always tell MP3 from uncompressed apart, no matter what encoder, no matter what encoder settings, no matter what source material. And they claim, that's because of their extraordinary ears. The sad truth is: The opposite is true. Their ears are not extraordinary, they are in fact broken. See, the psychoaccoustic model was developed for people with a normal working acuesthesia (= sense of hearing). Of course, some people hear better, others hear worse but the model was designed for people with "healthy ears". If your ears are damaged because you listened too very loud music as a teenager or if you were born with a hearing deficit or if your hearing sense was attacked by some illness, the model will simply not fit your ears. The way these people hear music is different to the rest of us, similar like color blind people see the world in a different way than those with standard vision.

This may not be a problem in daily life and these people can enjoy music just alike but as they may not hear certain frequencies, the model does not model their sense of hearing at all and in that case the compression based on that model will fail for them every time. E.g. frequencies that we cannot hear as other loud frequencies nearby cover them aren't covered for someone who cannot hear that other nearby frequency at all. So this person would hear the frequency that we don't according the model and if that frequency is dropped, the person will notice while we won't. So if someone can always hear the difference, this means nothing unless his ears got professionally tested and no hearing defect has been detected. A lot of musicians have defects because of too loud stage music causing permanent damage and that's why so many musicians claim that MP3 is crap and sounds horrible. Well, it does sound horrible but only to their ears.

To get an idea for the answer you can use a VBR encoder like LAME and use it as best quality (-V 0). At that level LAME tries its best to preserve every signal it considers hearable by human ears according to its model. So whenever a frame is encoded with less than 320 kbit/s, then no more than the bitrate of that frame was required to preserve all these signals. Here's a sample output of running such a command:

Note however that the example above was LAME trying to be as lossless as possible. Most people neither have ears good enough, nor equipment good enough to require such a level of quality. For the vast majority of people and home equipment, -V 2 should be perfectly transparent. So let me repeat that test again with that option:

And now you can see that to reach that level of quality, 192 kbit/s is in fact enough most of the time, often even 160 kbit/s is. There are only a few cases where LAME hat do go up to 224 and only very little where it had to go all the way up to 320 kbit/s.

One of the difference between -V 0 and -V 2 is that the first one considers all frequencies as important whereas the later one only frequencies below 18671 Hz, which is one of the reason why it has to encode less signals. Yes, human beings can hear ut

To get an idea for the answer you can use a VBR encoder like LAME and use it as best quality (-V 0). At that level LAME tries its best to preserve every signal it considers hearable by human ears according to its model - it aims to be perfectly transparent to all listeners for all source materials. So whenever a frame is encoded with less than 320 kbit/s, then no more than the bitrate of that frame was required to preserve all these signals. Here's a sample output of running such a command:

Note however that the example above was LAME trying to be as transparent as possible without just always using 320 kbit/s for file size reasons. Up to today, nobody has proven with an ABX test, that he can tell the difference between 320 kbit/s CBR and LAME -V 0 VBR MP3; which does not mean that people couldn't tell the difference between MP3 and uncompressed! It only means that when -V 0 was not transparent, CBR 320 kbit/s was not transparent either.

Yet most people neither have ears good enough, nor equipment good enough to require such a level of quality. For the vast majority of people and home equipment, -V 2 should already be perfectly transparent. So let me repeat that test again with this option:

And now you can see that to reach that level of quality, 192 kbit/s is in fact enough most of the time, often even 160 kbit/s is. There are only a few cases where LAME had do go up to 224 and only very little where it had to go all the way up to 320 kbit/s. So if I had limited encoding to 192 kbit/s, that would still have produced perfectly transparent frames to most listeners most of the time, just not to all listeners all of the time.

One of the differences between -V 0 and -V 2 is that the first one considers all frequencies as important whereas the later one only frequencies below 18671 Hz, which is one of the reason why it has to encode less signals. Yes, human beings can hear up to 20 kHz, but usually only small children. Already most teenagers cannot hear frequencies that high anymore and rarely any grown up adults, no matter how good their ears, no matter how good their equipment, so cutting these very high frequencies usually makes no audible difference for almost all listeners, unless these are young children and they will hardly care if these frequencies are missing in whatever source they are listening to. So yes, something is definitely missing here in the audio signal, regardless how many bits were used, but people who say they can tell first need to prove that by an ABX test and if such a test is made correctly, they usually cannot.

As for the psychoaccoustic model, one of the reason why AAC can achieve much better quality at the same bitrate or the same quality at a lower bitrate is because it uses a better and more fine grained model and encodes signals in a more fine grained manner, too, as one without the other one would make little sense. It would be possible to make a better model for MP3 but the format lacks the capability to make use of it. And it would be possible to design a model that works better even for trained ears but the result would only be encoders that produce bigger MP3 files in VBR mode with no benefit to the vast majority of users as they wouldn't hear any difference.

As a closing word, let me point out a rather sad fact. There are in fact a bunch of people that can always tell MP3 from uncompressed apart, no matter what encoder, no matter what encoder settings, no matter what source material. And they claim, that's because of their extraordinary ears. The sad truth is: The opposite is true. Their ears are not extraordinary, they are in fact broken. See, the psychoaccoustic model was developed for people with a normal working acuesthesia (= sense of hearing). Of course, some people hear better, others hear worse but the model was designed for people with "healthy ears". If your ears are damaged because you listened too very loud music as a teenager or if you were born with a hearing deficit or if your hearing sense was attacked by some illness, the model will simply not fit your ears. The way these people hear music is different to the rest of us, similar like color blind people see the world in a different way than those with standard vision.

This may not be a problem in daily life and these people can enjoy music just alike but as they may not hear certain frequencies, the model does not model their sense of hearing at all and in that case the compression based on that model will fail for them every time. E.g. frequencies that we cannot hear as other loud frequencies nearby cover them aren't covered for someone who cannot hear that other nearby frequency at all. So this person would hear the frequency that we don't according the model and if that frequency is dropped, the person will notice while we won't. So if someone can always hear the difference, this means nothing unless his ears got professionally tested and no hearing defect has been detected. A lot of musicians have defects because of too loud stage music causing permanent damage and that's why so many musicians claim that MP3 is crap and sounds horrible. Well, it does sound horrible but only to their ears.

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Mecki
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Are there any audible differences? Yes, certainly but not a lot.

Keep in mind that MP3 cannot guarantee transparent compression at any bitrate. Not even at 320 kbit/s every song or sound effect will be perfectly transparent. There are some unfortunate sound combinations that just cause pretty much every MP3 encoder to fail even at 320 kbit/s.

The question is: Are there are lot of cases where the encoder will fail at 192 but not fail at 320 kbit/s? And therefor you have to take a look why an encoder can fail in the first place.

An encoder will usually not encode a signal it deems "unhearable" according to its psychoaccoustic model, no matter what the bitrate is, as that would only waste bits for nothing. The problem is: There is no perfect psychoaccoustic model in the world! All existing models have been created by making AB or ABX tests with voluntary testers and are thus only as good as the hearing capabilities of the test persons as well as the quality of the equipment used for testing. So if the model is flawed, it may throw out signals it should have kept. And even it is not flawed, it only models the average listening capabilities of the average listener. Yet there are people with listening skills far above average, too. If the model is causing the problem causing the audible difference, raising the bitrate may not help much or may not help at all, as the encoder may still throw away a signal it should have kept.

However, an encoder may also fail if it knows that a signal is hearable, yet cannot encode it as it ran out of bits. At some point the encoder has to also throw away hearable signals as the amount of signals it can store is limited because the bitrate is limited. Encoders usually try to be clever and throw away the signals that should be least noticeable to be missing but even if one of these are missing, some people may be able to notice that. And if that is the problem causing the audible difference, then of course raising the bitrate will improve the situation.

So the question is: How often does an encoder has to throw away important signals it couldn't encode at 192 and that it could have encoded just fine at 320 kbit/s?

To get an idea for the answer you can use a VBR encoder like LAME and use it as best quality (-V 0). At that level LAME tries its best to preserve every signal it considers hearable by human ears according to its model. So whenever a frame is encoded with less than 320 kbit/s, then no more than the bitrate of that frame was required to preserve all these signals. Here's a sample output of running such a command:

Encoding as 44.1 kHz j-stereo MPEG-1 Layer III VBR(q=0)
    Frame          |  CPU time/estim | REAL time/estim | play/CPU |    ETA 
  7800/7800  (100%)|    0:03/    0:03|    0:04/    0:04|   57.172x|    0:00 
 32 [   1] *
 40 [   0] 
 48 [   0] 
 56 [   0] 
 64 [   0] 
 80 [   0] 
 96 [   0] 
112 [   0] 
128 [   7] %
160 [  28] %
192 [  13] %
224 [ 829] %%%%%%%%%%%%%%%%%%%%%***
256 [4192] %%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%********************
320 [2730] %%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%***************************
-----------------------------------------------------------------------------------------------------------------------------------
   kbps        LR    MS  %     long switch short %
  274.4       77.1  22.9        85.2   8.0   6.8

As you can see, LAME was convinced that most of the time more than 192 kbit/s are required. If you had limited the MP3 to just 192 kbit/s, then most of the time hearable signals would have been dropped. Also notice that only in case of a 320 kbit/s block, LAME may have been running out of bits, as in all other cases, it could just have increased the bitrate if that was the case but apparently in some cases 256 or 224 were enough and in a few cases even 192 kbit/s were.

Note however that the example above was LAME trying to be as lossless as possible. Most people neither have ears good enough, nor equipment good enough to require such a level of quality. For the vast majority of people and home equipment, -V 2 should be perfectly transparent. So let me repeat that test again with that option:

Encoding as 44.1 kHz j-stereo MPEG-1 Layer III VBR(q=2)
    Frame          |  CPU time/estim | REAL time/estim | play/CPU |    ETA 
  7800/7800  (100%)|    0:03/    0:03|    0:03/    0:03|   63.878x|    0:00 
 32 [   1] *
 40 [   0] 
 48 [   0] 
 56 [   0] 
 64 [  27] %
 80 [  17] %
 96 [   4] %
112 [   5] %
128 [  14] %
160 [1554] %%%%%%%%%%%%%%***********************************
192 [3849] %%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%%*************************************************************
224 [1167] %%%%%%%%%%%%%%%%*********************
256 [ 830] %%%%%%%%%*****************
320 [ 332] %%%%*******
-----------------------------------------------------------------------------------------------------------------------------------
   kbps        LR    MS  %     long switch short %
  201.7       41.4  58.6        85.2   8.0   6.8

And now you can see that to reach that level of quality, 192 kbit/s is in fact enough most of the time, often even 160 kbit/s is. There are only a few cases where LAME hat do go up to 224 and only very little where it had to go all the way up to 320 kbit/s.

One of the difference between -V 0 and -V 2 is that the first one considers all frequencies as important whereas the later one only frequencies below 18671 Hz, which is one of the reason why it has to encode less signals. Yes, human beings can hear ut