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11
General Audio / Re: 89 db replaygain too much for classical music
Last post by timcupery -
The dramatically easier way I've found to do this is to encode to +1.5 or +3 or +4.5 (etc.) db above the target volume, and then use fb2k or mp3gain to lower the resulting file by that same amount. I'm doing this with FB2k, and this may engage some anti-clipping algorithm that I'm not aware of and for which I can't find a switch. I haven't tried it with commandline or LameDropXPd to see if baseline LAME encoder does the same thing.

Here's an example of a song that I'd encoded years ago (original was with LAME 3.92) that I just left much quieter than I'd prefer it to be. Recently I scaled it up to volume that sounded right (it's one of these weirdly-mastered indie songs where RG really misdiagnoses the perceptual loudness) and used the method I described above to get the same RG volume, without any clipping. This peak-lowering only decreased the RG value by 0.12 dB.

Note: I'm unable to ABX-differentiate the no-clipping version from the WAV, even though I did a fair comparison with scaling them down to the volume where the WAV isn't clipping, and just turning up my speakers.

Attached are screenshots showing the RG and peak values, as well as wav exports of those two mp3s (the 100% one has clipped peaks that go up to 189%)
13
General Audio / audio editor recommendations (I'm still using EAC's sound processing)
Last post by timcupery -
I got started using the built-in sound processing from Exact Audio Copy, in the fairly early days of HA. I still use it and am comfortable with it, although I'm older and have a family so I don't do much anymore.

EAC's sound processor is limited to normal CD-rip (44.1 16-bit stereo wav) files, and I'm increasingly getting lossless files with higher samplerate or bitdepth (e.g., buying remastered albums through Bancamp).

I've thought for over a decade that I should switch audio editors, but haven't put time into figuring out something that can handle a wider range of files.

Things I like about EAC's sound processing:
* basic wavform display, can zoom in horizontally (time interval) and vertically (wavform height)
* can click on individual samples to see/edit the exact value
* interval bars for CD audio frames (588 samples, 75 per second at 44.1khz)
* options such as "select peak range" and "scale selection"
* doesn't change any sample values unless I tell it to (which I expect to be standard behavior for handling any lossless file)

Audacity is the audio editor I'm most aware of, and I know it handles a wider range of file specifications and types. Will it (or some other app) do the things I'm looking for that EAC's editor does?
14
General Audio / Re: 89 db replaygain too much for classical music
Last post by timcupery -
I don't often encode classical music, but I'm aware of this problem and have encountered it from time to time in rock/pop music (hardly ever with more-recently-produced-or-remastered albums, which tend to be more compressed).

My old solution, when I was in early 20s and single, was to manually select each offending peak (from where that the amplitude line crossed 0 dB, to where it crossed back again, so peak about in the middle of selection) and scale that whole selected area so the peak was compatible with my target scaling on encode. As one might imagine, this was quite time-intensive, and the sort of thing a programmed loop or some other algorithm could have handled much more efficiently.

The dramatically easier way I've found to do this is to encode to +1.5 or +3 or +4.5 (etc.) db above the target volume, and then use fb2k or mp3gain to lower the resulting file by that same amount. I'm doing this with FB2k, and this may engage some anti-clipping algorithm that I'm not aware of and for which I can't find a switch. I haven't tried it with commandline or LameDropXPd to see if baseline LAME encoder does the same thing.
15
General Audio / Re: preferred/best method for encoding 48khz FLAC --> 44.1khz lossy
Last post by timcupery -
LAME can, by commandline, handle resampling internally while encoding. For my case, starting with a higher-samplerate input file, I would add
--resample 44.1
to the commandline

It doesn't appear to FB2k can tell LAME to do this, nor can LameDropXPd. I've usually tagged my files with RG album gain tags, and with commandline I may have to translate those into --scale X commands. For example, an album gain of -6.0 dB would translate into --scale 0.5
18
3rd Party Plugins - (fb2k) / Re: [fb2k v2] Playlist Attributes (foo_playlist_attributes)
Last post by cwb -
This is strange to me:

I have three playlists that I use Playlist Attributes "Edited" DSP Settings with.

The rest of the playlists use the Global setting, which is no Active DSP entries.

The scenario:

Step One: I play a file in one of the three "Edited" DSP Settings playlists, I press "Stop", and then open "Preferences\DSP Manager" and no active DSP's are listed. (I take that to mean any Active DSP's have unloaded.)

Step Two: I then go to one of the Global (no Active DSP entries) playlists, and double-click a file to play it. The file starts playing, and I then open "Preferences\DSP Manager", and no active DSP's are in the list. (All is well).


But if I do Step One, listed above, and instead of Step Two, I go to one of the Global (no Active DSP entries) playlists, and select a track and add it to "Queue", and press "Play" to start playing the queue, the "Preferences\DSP Manager" list the Active DSP's that were used in the previously used (Step One:) "Edited" DSP Settings playlist.

Maybe I am doing something wrong that is causing this to happen. Or some other component I have installed is causing conflict? Or maybe nothing is really wrong and I am just thinking something is?