Few weeks ago I bought my first MP3 player, just to have some music on the road (bike, train). It sounds awful. Really bloody cracky and everything you could ever imagine. So, I thought my ears were too sensitive or something, and started measuring.
So, just made a quick frequency response of my MP3 player versus a cable between sound card in- and outputs:
Soundcard:
MP3 player:
I didn't know it could be this awful and I am scared of even approaching a reasonable guess for its THD...
So, my question:
What do these guys put in a device like this to make it this horrible???
So, just made a quick frequency response of my MP3 player versus a cable between sound card in- and outputs:
Soundcard:
An externally hosted image should be here but it was not working when we last tested it.
MP3 player:
An externally hosted image should be here but it was not working when we last tested it.
I didn't know it could be this awful and I am scared of even approaching a reasonable guess for its THD...
So, my question:
What do these guys put in a device like this to make it this horrible???
Limhes said:
Psychoacoustic lossey compression is a nonlinear process - the “distortion” results are readily understood as expected from the details of the compression algorithms
By exploiting a variety of masking phenomena, nonlinear loudness perception and "spending" bits only to encode what features make a difference in what people hear better audio compression schemes can "toss out" >75% of the data in a 16/44.1 CD with a vanishingly small number of people able pick to out the difference in blind testing
see the tech papers section of
http://www.mp3-tech.org/
I understand, but aren't you talking about MP3 (or lossy) compression in general? I know how these work, generally, involving acoustic masking.
I used an MP3 file (the same!) in both cases. The distortion figures are from my portable MP3 player only! Now, this player just takes the files I give it and plays them, but why would they ever distort these files even more? You think it's a loudness algorithm inside the device?
I used an MP3 file (the same!) in both cases. The distortion figures are from my portable MP3 player only! Now, this player just takes the files I give it and plays them, but why would they ever distort these files even more? You think it's a loudness algorithm inside the device?
I suspect some automated tests could look bad based on a mls/noise or multisine test where the compression algorithm is busy exploiting psychoacoustics to remove "unhearable" features from the test signal
A slow single sine sweep passed through the mp3 encode/decode chain should give a flat frequency response, likewise single tone test sine should show no added harmonics - within the arithmetic resolution limits
your second plot shows classic square wave harmonic structure - as a guess you may be severely mismatching levels and clipping badly
I'd expect few portable mp3 players would saturate a typical soundcard line in - check to see if you're not accidentally configured for mic in or just reduce the test tone level to see if you get an abrupt improvement as you quit clipping
A slow single sine sweep passed through the mp3 encode/decode chain should give a flat frequency response, likewise single tone test sine should show no added harmonics - within the arithmetic resolution limits
your second plot shows classic square wave harmonic structure - as a guess you may be severely mismatching levels and clipping badly
I'd expect few portable mp3 players would saturate a typical soundcard line in - check to see if you're not accidentally configured for mic in or just reduce the test tone level to see if you get an abrupt improvement as you quit clipping
sorry for threadjacking.
I_F,
http://www.arrl.org/w1aw/morse.html
the mp3s were played back on xmms on linux. i'm going to try some other players tonite. tnx.
I_F,
http://www.arrl.org/w1aw/morse.html
the mp3s were played back on xmms on linux. i'm going to try some other players tonite. tnx.
I don't think there is a bitrate low enough to produce that kind of distortion. You can define a sine wave accurately with a few bytes.
That waveform just isn't an accurate representation of what is coming out of the MP3 player. At least not unless one of the sound-enhancing effects is "Fuzzbox". You might as well ask why a Britney Spears MP3 doesn't sound like Kiri Te Kanawa.
That waveform just isn't an accurate representation of what is coming out of the MP3 player. At least not unless one of the sound-enhancing effects is "Fuzzbox". You might as well ask why a Britney Spears MP3 doesn't sound like Kiri Te Kanawa.
jcx said:
I guess I won't make this kind of mistake, but here it goes: I set the MP3 player to max volume (this is the worst thing to do, but that was my intention) and the input to min gain (external sound card), using line in of course. Then, I set Audition to recording, and advance the gain control until I get a normal signal level. Then I restart recording and make an FFT.
Edit: aah, now I know what you're saying. The max volume could be too big as input line level. I'll check it out...
janneman said:
Hmm, does it work that way? But then it must be the decoding, since that's what happens inside the MP3 player. Why would an algorithm do that? Why not get the best reproducible quality of music out of the file??? This doesn't sound very logical...
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