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The best advice I can give is: Don’t panic! ;)

As well tested as LAME is it’s extremely unlikely that V0 is anything but perfectly transparent – maybe you’ll find problematic samples, but they’ll be few and far between.

But there’s nothing wrong with doing some ABX tests yourself to get back your peace of mind. If you’re not able to hear a difference between the original and the encoding in a double-blind test done in your usual listening environment(s) with the music you listen to all the time, then there isn’t any point in switching formats, is there?

If even that’s not enough to combat the paranoia, then consider solving the problem once and for all by switching to a lossless format (e.g. FLAC).
Silly comment but if
Do you make music transcription? Some audio processing plugins manipulate the phase of stereo signals to emphasize or minimize some instruments panned on a specific position to make transcription easier, the most common one is the so-called "vocal removal" technique which assumes vocal is mono and centered on the stereo field.

With mp3 or some other lossy codecs such techniques could yield poorer performance.

For only that issue I have found that AAC-LC is the best option by far from any other, but I repeat, only for that issue.

If your don't have any other of the common codec-switch issues like running out of storage on that particular device, need a more efficient codec for given (lower than transparent) bitrate or compatibility, then I don't see why you would need to re-rip everything from your catalog if everything is already transparent to you, you'll gain nothing.
I had this idea too, I'm glad I wasn't the only one.
Well, atleast 90% of all Top40 tracks you listen to the radio, if you create a duplicate in a DAW and you invert it's phase, then match the first verse with the second verse and the first chorus with the second chorus, pretty like 75% of the content gets cancelled out in the verses (the rest will be mostly some reverb and the vocals) and the chorus will be mostly cancelled out as a whole. I bet that a potential codec that could take advantage of such thing (lossless or lossy, both could get potential advantage) would have one problem. When things start to get out of phase and they are not identical anymore it makes it inefficient and hard to find a way to compress uniformly. I mean you could have a typical Top40 pop track that goes like verse1-chorus-verse2-chorus-bridge-chorus that could be compressed to crazy small files (lossless or lossy) with amazing efficiency (percivable quality or compression ratio) and then a piece of classical music that most possibly has nothing that it's phase is a douplicate of an other part and it will compress poorly. Still though, there are so many video codecs that have tuning for different types of inputs to them, creating a long term prediction subcodec/codec that is based on the idea that you may find the same thing again on the file with potentially minor differences wouldn't be as much of a bad idea as alot of people will try to claim that it is.

Your only enemies are processing power, long encoding times (because of looking waaay ahead to the file) and the fact that you need a coder that is very smart when he writes code, he must definitely be a music producer to "get it".
Do you make music transcription? Some audio processing plugins manipulate the phase of stereo signals to emphasize or minimize some instruments panned on a specific position to make transcription easier, the most common one is the so-called "vocal removal" technique which assumes vocal is mono and centered on the stereo field.

With mp3 or some other lossy codecs such techniques could yield poorer performance.

Sometimes you may also want to slow the song down to pick up some fast notes, lossy codecs could also yield poorer performance.

However, in my experience, if a stereo mp3 is transparent (audibly lossless), it will still transparent when only one channel is being played.
Sure, the chorus of a particular song may have the exact same musical notation and the same lyrics, but no band plays perfectly according to the notes, 100% every time. There will always be variation, some of it is deliberate, some of it isn't. If you take that away, and basically just play a recording of the chorus every time you get to that part of the song, it'll make the songs lifeless, generic and boring....
If a (compressed) MIDI file represents the core musical information of a song, then might not "musically guided" loss represent sensible rate:distortion compromises? While repeated musical sections might be numerically different (due to noise, performer involuntary variation or performer conscious variation), it sounds like an interesting axis to work in.

I know that scores of e.g. Bach music may not be available from the composer himself, rather, some musically gifted soul listening to Bachs performance went home and transcribed the music from memory. While I find that capability amazing, I guess that there will be errors. Hopefully, the "essence" of the music survived this operation.

Perhaps because I am interested in both dsp and music, I am intrigued by the idea of lossy codecs being able to analyze a piece of music in a musically sensible manner (i.e. waveform to score) perhaps using some (algorithm du jour) machine-learning mechanism, then figuring out what matters the most to (to e.g. someone coming from a western musical tradition), and how to spend bits most wisely.

The comparision to video codecs is interesting. AFAIK, they don't even have an explicit model of our vision (unlike audio codecs), and when they track "motion" across temporal frames, they will often find "apparent" motion that does not correspond well with actual motion. I.e. they will pick up "something" that allows them to encode the residual with fewer bits, but nothing like a plausible optical flow type modelling.

Hi guys :) I could do with a little advice, please…

For the last 15 years I've happily chosen to encode my music with LAME mp3 (V0). I've always considered this perfect for me for a few reasons…

My criteria
  • I am fussy about quality - but not to extremes
  • I'm only a casual listener, with cheap earphones, I'm not an "audiophile" when it comes to listening to music
  • I enjoy music, but ultimately my collection is not terribly important to me
  • I'm happy as long as the music sounds like original most of the time - I don't mind the occasional 'imperfection' or 'glitch'
  • I'm not fussy enough to care about the whole "subconscious perception" issue
  • I do have a very large music collection, and even though hard drive space is cheap, I don't like the idea of how much space lossless would take up
  • Another key factor for me is that I'm a "minimalist" and find it satisfying to know that my music files take up as little space as possible, with all the unneeded data stripped away

So LAME mp3 has always been fine for me, on this basis.

But lately I've been wondering if it might be time to switch codec? Maybe there's something much better than mp3 these days? Perhaps ogg or something? I really known nothing about the other lossy formats.

But in particular, something else has been bugging me about LAME mp3… As a musician myself, I have a real interest in how all the parts in the track are mixed. So, while normal listeners will just focus on the lead vocal and main melody - my ears are more sensitive to "every single part of the mix" - I will often be paying attention to individual instruments in the background, or individual notes in chords - things that are perhaps really subtle in the mix - two instruments blended together, or one very quiet instrument in an orchestra hard-panned to the far right, just about audible in one ear. These are details that most people wouldn't care about. And then sometimes I will listen just to an isolated channel (left or right) just to hear what is going on in each channel.

So I guess I'm asking: Is LAME V0 good enough for me? I'm not obsessively fussy about the music being "perfect", but I do want to be able to hear all the parts and all the harmonies, in all the mix, across both the channels - including all the subtle nuances of every instrument (something a regular listener may not care about).

And can LAME V0 be trusted for playing just the left or right channel in isolation? Or does it start to break down when you do that? (To my ears, LAME V0 does sound identical to the original WAVs, but I haven't done enough testing to put any confidence in my own conclusions.)
I am somewhat duplicating a post that I wrote in the foobar forum, but I have a reason : the problem is still not solved and issue persists with another player (Musicbee). I suspect it has nothing to do with Foobar - which worked well until the problem arose.

I have drop-outs only when sending the music signal through the usb cable to my DAC (Teac UD501), otherwise, the music has no drop-outs when played through the computer speaker.
I have tried increasing the buffer size, stoping all other software, and even unplugging the ethernet change. I have a Dell Vostro Intel I5, 4 Mo R, Nvidia Gforce GT330M.

Any idea thaty could help ? Thank you.
UMG doesn't use different mastering for their watermarked stuff. I am completely boycotting UMG content because of this reason. It defeats the whole purpose of offering your content in lossless form. Go unwatermaked or go home period. No, it's not okay; never was, never will be.
At least you have two choices!

CD: intentionally crippled mastering
Download: bloated hi-rez files with better mastering + potential watermark