![]() ![]() The only thing I hesitate about reading your posting, is if there indeed should be (possibly selectable) indicators as: The setting of -21 LUFS means the humanly perceived loudness of the whole track has a loudness of 21 LU's below the maximum. With my proposal of using LUFS you could explain: Setting the slider to -3dB would result in a loudness that might be perceived as 86dB loudness, which is in itself is an incorrect statement, since if you would turn down your volume knob all the way down, it would be 0dB loudness. With the slider set to -3dB currently, you would have to explain it by saying: well, there was an old standard, which has never been official acknowledged by anybody, which MusicBee no longer uses, that at a certain moment advised on a certain perceived loudness level (which was calculated through a by now obsolete and inferior model). My suggestion is of course also to make it easier to grasp for new users. Hopefully this won't be attacked as it being 'to technical', since it is just factual, and the matter of fact is that we are dealing wit a very technical (background) of the matter. Since all the newer standards (ie: ITU-R BS.1770, EBU R128, ATSC A/85 and ReplayGain 2.0) are using LU & LUFS as the preferred measurement, then it would also make sense to change MB.Īs MB currently uses R128gain in ReplayGain compatibility mode, the scale should center on -18 to identify that this is the starting point with no manual adjustment.Īs the author of R128gain has released an updated version (now as BS1770Gain) MB3.0 also should take advantage of this and update as well.Īs the newer version supports switches for R128, ATSC and ReplayGain compatibility, MB3 could have selectable options for which reference loudness level you want to use and would automatically redraw the adjustment scale centered around that selected point īS1770Gain also has the advantage of also writing a loudness reference and algorithm tag I am very open to contributing and funded objections, so if you would reconsider that would be fine by me. I believe you were not just giving your personal opinion, but attacking my proposal with incorrect statements. I thought I'd mention it in case that bothers you.Replace dB with LUFS, and have a 'loudness' tag. Foobar2000 said the track gain is -1.55dB, so in that case the difference between scanning algorithms is around 1.5dB. ![]() MP3Gain had saved a TrackGain of +0.06dB to the tag. The track I used for converting was originally scanned with MP3Gain so I scanned it again before converting it. Mostly they tend to agree to within about 1 or 2 dB, if memory serves me correctly. and Foobar2000 uses the EBU R128 scanning algorithm rather than the original ReplayGain algorithm, which is probably a bit more accurate, so the result can be a little different if you scan with MP3Gain. ![]() FLAC first, AAC second in the screenshot below. Lossy codecs tend to deviate by a small amount, but that's what lossy codecs do.Ĭonverting to 95dB. If you convert to a lossless format and scan the output files you should end up with exactly the volume you specified, aside from any deviations that might be caused by clipping or limiting. I adjust the volume losslessly if it's possible without clipping, but for conversion I created a preset with a limiter DSP in the conversion chain. I chose 93dB as the number of tracks that'd clip was still reasonably low. She uses it to teach line dancing in a hall with a small sound system so I have to squeeze as much volume out of it as I can. I have the job of putting the music on my mother's MP3 player. The player isn't involved in the conversion process, although they both decode with ffmpeg. ![]() The converter uses command line encoders and will decode/output to 32 bit float if an encoder supports it. If it doesn't have to be lossless, then yes foobar2000 will do it accurately. The 1.5dB step limitation is an MP3 limitation. ![]()
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