with 2): For the very reason you mentioned: I want an exact reproduction, in terms of silences and even hiss (it can add that special flavor, think old jazz records). Silence is encoded at 32 kbps, in MPC even lower, so there's no problem.
with 3): You obviously strive to boost the volume as much as possible (without clipping), however, the perceived loudness doesn't depend on the peak volume (compare dynamically compressed music with classical music). While peak normalizing usually adds some 3 dB noise, the loudness gain is maybe <1 dB with today's music productions. And don't forget that the music has already been peak-normalized in the studio.
with 5): Why do you always want to increase volume? Many playback systems start to have problems at -3 dB below FS already. The reference volume in ReplayGain was introduced to have a safe, uniform volume level across the board. You put this ad absurdum. But i'm sure, one day you'll discover the great benefit of ReplayGain...
As for the hiss, I should be clear that I'm only talking about the hiss at the very end of a track, where the song itself is over but the master tape hiss is still audible.
Are we talking about 2 different types of dithering?
BTW - what does FS stand for? And what playback systems are you aware of that have problems as amplitude approaches 0db?
to crank up the volume to abnormal levels just to hear it.
3) no, no, no (this is my opinion, if mp3 is your goal, as you have written).
4) umm, --alt-preset standard with addition -V1 is nowhere recommended. There are no test sessions made with this. In fact, the alt-preset was optimized as it is, so you do only cause trouble by adding NOT RECOMMENDED SWITCHES. For higher or lower quality or filesize, choose recommended presets from list !!
5) check for clipping (yes, here, not earlier like in 2, normalizing the mp3 is lossless by mp3gain, so it is better, and otherwise you would have double work, here in mp3gain you can adjust volumes much better than in step 2.)
I don't need 8 steps...
hmm, probably your misconception of mp3
So, the question comes down to this: Is it better to adjust the .wav before encoding where the adjustment increment can be virtually unlimited but introduce rounding errors to the original sample values or adjust the MP3 file after encoding where the adjustment increment is rather coarse, but applied to a file created from a "true" source?
Here's the file i mentioned before: overloadtest.flacCheck when your soundcard distorts. Use headphones if you can, and don't turn it up too loud.
If you have been around for a while, the forum in general recommends using mp3gain based on the philosophy of not touching the original as much as possible, and I concur.Another reason to use mp3gain is that you can be sure that the result will not clip. If you do your own normalizing before hand, you have absolutely no idea how much you need to attenuate by to prevent clipping.
Regarding -v1 and -v2:There are many things to adjust which can affect the quality of a VBR encoding, and -v is one of them. -v mainly deals with the position of the ath curve iirc, and is not really the optimal way of scaling VBR presets although it is a quick and easy way. During testings, Dibrom found that going below -v2 is redundant so that is probably around where the real ath lies. In most cases you get larger bitrates but the quality would not be noticable, and the bits are probably better spent in other ways. In some cases, especially when you hit the 320kbps frame limit you may get lower quality where bits are not optimally used.
Yes, I'm aware of that perspective, and, in general agree with it. I need to do more research into the effect the rounding errors have on the samples when applying the adjustment to the .wav file. I've found, in limited testing to be sure, that an absolute peak normalization of -2db prevents clipping when encoded. Further testing might reveal this number can be a bit less, as well as that there are tracks where even this isn't enough, although it passed muster on one that was extremely loud and compressed the whole way thru.
Would you mind explaining in a bit more detail what I'm looking for and how to do it
This depends on settings used.-aps should clip very rarely
Just listen to it. You shouldn't hear the differential tone at 3.7 kHz. If you do, your soundcard distorts. Bad soundcards already distort at -6 dB, better ones maybe at -2 dB. The sample goes from -12 dB to 0 dB.
After turning it up, I still don't hear anything at first, and then it's a constant sound that gets louder until it stops.
Looking at the analysis from the CD I used to test various settings, MP3Gain reports that 11 out of 13 songs are clipping when encoded with --a-ps and --aps-V1. When encoded at --a-ps-V0, it drops to 9 out of 13. And that's based on a straight encode of the .wav with absolutely no adjustments made to the source before encoding. Yes, it's a loud CD of rock Xmas music that seem to be from various sources and not much time was spend mastering them together, and it's just one CD. But that's the datapoint I have.
Can I use an older version or does this FAQ need to be re-written?