Hello,
I have a MP3 file that I would like to analyse using Audacity. The file was recorded with 320 kbits/s, and when I try to import it, I get the message “forbidden bitrate value”. I tried to change the quality in preferences, but it did not change anything.
I finally converted it in a lower bit resolution, but I fear a loss in quality. Does anyone have an idea to help me and allow me to use my “normal” file?
Thanks a lot
LV
Testing on W10 with 2.4.2 and the latest alpha test build I have for the upcoming 3.0.0 I can import a two hour stereo MP3 that was encoded at 320kbps - created and sent by a friend of mine (he created it with Audacity 2.4.2).
So works-for-me 
WC
Hum, strange, cause I’m using also the 2.4.2 version…
What does MediaInfo say about that file?
The error message is probably wrong… It may not actually be MP3 or there might be something else wrong with the file.
I assume you can play the file (with Windows Media Player or whatever you normally use)?
I finally converted it in a lower bit resolution,
How did you do that? If that worked, “converting” it to 360kbps would probably work too but any MP3 re-compression will degrade the quality (but it probably won’t affect your “analysis”). Try converting it to WAV. (Audacity would have to decompress the MP3 anyway.)
If you don’t have a tool to do that try [u]TAudioConverter[/u] or [u]Kabuu Audio Converter[/u] to make a WAV file.
Quite interesting! It actually wasn’t really MP3… Why?? Why if the parameters of my recorder device are MP3 output, do I get MPEG (I don’t even know what that format is ^^) But that’s another question…
I used the “HD Video Converter Factory” software. I’m gonna try to convert it to 360 kbps as well, to see.
Okay, I’m gonna try in WAV. I will need the FFmpeg library don’t I?
Thanks for all these advices!
Perhaps you could give the answer to my first question: “What does MediaInfo say about that file?”
Sorry, it said that the format was MPEG
What’s the rest of the info? MediaInfo gives a lot of information, including sample rate, bit rate, number of channels …
Just copy and paste the whole lot into your reply.
Okay, I got that:
MediaInfo report of “2020_11_01.MP3” :
General
Complete name : 2020_11_01.MP3
Format : MPEG Audio
File size : 3.33 MiB
Duration : 1 min 27 s
Overall bit rate mode : Constant
Overall bit rate : 320 kb/s
Audio
Format : MPEG Audio
Format version : Version 1
Format profile : Layer 3
Format settings : Joint stereo
Duration : 1 min 27 s
Bit rate mode : Constant
Bit rate : 320 kb/s
Channel(s) : 2 channels
Sampling rate : 44.1 kHz
Frame rate : 38.281 FPS (1152 SPF)
Compression mode : Lossy
Stream size : 3.32 MiB (100%)
That all looks OK so it IS an MP3 file but there’s something (unknown) that’s “non-compliant” with it.
Okay, I’m gonna try in WAV. I will need the FFmpeg library don’t I?
No… FFmpeg isn’t (normally) used for MP# and the Audacity FFmpeg library won’t help if Audacity can’t open it. 
If HD Video Converter Factory can’t make a WAV, try one of the suggestions I gave you.
Sorry if this is a stupid question, but what is the difference between MP3 and MPEG?
“MPEG” is a group of formats that includes “MP3” (see: https://en.wikipedia.org/wiki/Moving_Picture_Experts_Group).
As DVDdoug wrote, the info from MediaInfo looks OK.
The error message “forbidden bitrate value” comes from “libmad” which is the MPEG decoding library that Audacity uses.
Previous versions of Audacity ignored errors from libmad, but that was causing some problems, so Audacity 2.4.2 enabled libmad error checking. Unfortunately, since doing that, we have found that a lot of MP3s have minor problems and will therefore not import into Audacity 2.4.2. The next Audacity release will still have some error checking of MP3s, but it will be more relaxed than in 2.4.2 so that most minor errors will be ignored.
The best thing to do would be to use another app to decode the MP3 file to stereo WAV and then you will be able to import the WAV into Audacity. The sample rate of the WAV should be 44.1 kHz (44100 Hz) and it should be stereo.
If you don’t have an MP3 decoder, there’s a good one built into foobar2000.
Thanks a lot!
I tried to convert it into WAV with the HD Video Converter Factory software that I already use, and it worked, so I think I’m gonna continue like that. I could actually see a small difference of quality on the sonograms, between the 128 kbps MP3 and the WAV.
Thank you Steve and DVDdoug for all your help, it has been very useful!!
I could actually see a small difference of quality on the sonograms, between the 128 kbps MP3 and the WAV.
Just FYI - Spectrums & spectrograms can be “interesting” and they can tell you something about what the compression is doing but it’s NOT the best way to judge audio quality. Often you can see a difference that you can’t hear.
MP3 is lossy compression so it will throw-away some information. Mostly, it tries to throw-away sounds that are masked (drowned out) by other sounds.
Often, the MP3 can sound identical to the uncompressed original in a [u]blind ABX test[/u]. And, if a 192kbps file sounds identical to a 320kbps file we can’t say the 320kbps file is “better” even though the 192kbps file is throwing-away more information. In both cases all of the samples are altered to some extent.
…It’s probably easier to make a compression algorithm that makes “pretty” spectrograms than one that makes good sound.
As I’m working mostly on sonograms, and not much on the hearing, it helps to have ‘pretty’ sonograms, but yes, maybe it was only a “feeling like better” ^^