And also because, well, I'm not sure if the same applies to ogg, but at least mp3 compression causes distortion at near-0dBv levels, so it's best to normalize to a bit less than 0dB, like -0.2dB. Mainly because you may need to do some post-processing steps, like normalization. you can pretty much just ignore them at this point, they will be useful in later versions though.ĭiiz wrote:Firstly, it's always best to import as wav, not ogg. Sample-exact and Anti-aliasing oscillators are features which haven't been implemented yet, so their presence is a bit confusing. If you plan on doing any post-processing, it's better to not oversample and just render in the higher samplerate instead, then do the downsampling manually later (same principle applies as with lossy/lossless formats, as explained above). Basically, using 44100 hz samplerate with 2x oversampling is practically exactly the same as using 88200 hz samplerate with 1x oversampling (except, in the second case, the resulting sound file will be 2x larger). Oversampling does what it says: it renders in a higher sample rate than specified, then resamples it down to the specified sample rate. In any case it's pretty safe to use the medium setting, as recommended. I think it currently only affects samples in the AFP (not sure though, will have to check). Interpolation is hard to explain, it's a mathematical algorithm used for resampling sound. But briefly, the same applies here: bigger is better. when you render in a higher sample rate, the calculations used for rendering the sound are more accurate, resulting in a better sound quality.īitrate only applies to ogg files, so we can ignore it. Generally, 96khz is a pretty good sample rate to export, even if you later downsample it to a lower rate. The higher the sample rate, the better the quality. It's how many times per second the sound is sampled to produce the waveform. Samplerate is the "resolution" of the sound file. Audacity is very good for this kind of post-processing. So I always import as wav, do the finishing touches in Audacity (may need some extra limiting/compression, as well as some slight equalization, in addition to normalization). With both audio and image editing, the principle is the same: always use lossless formats for editing and only convert to lossy as the very last step. If you export directly to ogg, you can't edit the resulting sound anymore (or you can but not losslessly, meaning the lossiness cumulates: it's kind of like taking a photo of a jpeg and then saving it again as a jpeg). (1.2.0 is also affected, but in a different way, and I have reason to believe that specific odd bug was fixed.)ġ.3 alpha makes it clear that the problem is a pitch difference.Firstly, it's always best to import as wav, not ogg. Their pitches are different Affected LMMS versionsġ.3 alpha. Export a wav file with settings set to a non-default oversampling.īoth sound more or less the same Actual behavior.Open the Piano Roll of this track and place a note there It creates an AudioFileProcessor instrument track.Open the drumsynth folder, drag in any of the samples.Open the sidebar and navigate to My Samples.Oversampling, on export, changes the output pitch of a ds (drumsynth) file note in an AFP instance. The last similar issue of pitch change with oversampling was for AFP files in general, and closed at the end of 2019.
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