G’day,
I have cleansed my collection of heaps of tracks lately and have some tracks that I need to re-purchase as the files I currently have are not up to scratch.
A thought was to find all the “low bit-rate (less than 256)” files and start with going through them to decide if I needed the track (in which case buy a good copy) or get rid of it once and for all.
I was hoping someone would be able to provide a reliable programme that will scan the “real” bit-rate as I feel I may have converted some low-res files to 320kbps when I was still an ignorant dumb-ass!
Alternatively, is there a way to delete the stored bit-rate and re-analyse?
Import a track, and then look on the left hand side of the waveform for a drop down menu (it will have the file name on it.)
Then select “Spectrogram.”
A true 320 kbps track will have frequencies at 20k.
If it cuts off a lot lower, then you have a transcode. Typically you’ll find it cutting off at about 16k, which means a 128 kbps MP3 was transcoded to 320 kbps.
Once you encode an MP3, that data is gone forever. There’s no way to get it back. That’s why most people recommend downloading only lossless (WAV, FLAC, AIFF) and then transcoding that to MP3s when needed, but keeping the original files for archival purposes.
Maybe. If the original source material had 20kHz, then a single encoding to 320kbps with a reasonable codec will likely preserve that content. Transcoding multiple times can reduce or remove high(er) frequency content.
LAME is well known for low passing the source material when encoding at or below 128kbps. BUT, if the encoding was done at 160kbps, or 192kbps, etc, then this signature might not be obvious. Other mp3 encoders (Fraunhofer) leave different fingerprints.
Overall, there is no “definitive” way to determine previous encoding bitrates of a sound file.
What exactly do you mean by transcoding multiple times? Like WAV > FLAC > ALAC?
That shouldn’t remove any content.
This is true, there are encoders that are problematic.
No perfect way, but provided he was just using something common like iTunes, that sort of transcode should show up, especially if it cuts off dramatically.
The lossless formats can include content up to half the sampling frequency, which is 22k for “CD quality.” That does not guarantee that the song was not low passed at some point in the original production process.
Your example was lossless to lossless conversion. That should be “clean,” at least in theory.
The conversion from lossy to lossless to lossy can introduce “artifacts” and can cause high and low frequency content to be compromised. The details depend on the details.
While most people will not “intentionally” transcode “a lot” of times…it can happen accidentally. Most music editors will expand a file into a “wav” format. If you open an mp3 in an editor and save as an mp3, every time there is a mp3 → wav → mp3 conversion. Edit a song a mp3 a few times, and you have done something like this:
Thanks for the input… It would appear that some idiot (me) had the bright idea of converting some wavs and aiff’s to 320kbps MP3’s but instead of doing just those; decided to do anything that wasn’t already 320… Like I said best intentions but done without the necessary knowledge… i.e. how did I think I 128 or 192 kbps MP3 would magically become a 320?
Sometimes it amazes me that I can walk! LOL I was hoping someone would have a solution but it looks like I have a LOT of work to do.
