For DJing you would be surprised at how low in bitrate you can go if you want to, but then: is space really an issue? Why are you using MP3 in the first place? Is it that it is the only tag-able format supported by a certain application?
Yes software, external players, compatibility and having 1000's of tracks, lossless just isn't practical. Like I mentioned I DO use strictly only Lossless, either WAV or AIFF for my 'proper' DJing, where I would only have around 20 tracks ready on a memory stick! (where I'm turning up to an event with high quality equipment that has been specifically set up with high quality sound in mind and people who know music and are expecting a great sound and musical experience) very different to my more regular djing playing 'chart favourites!'
Last post by czyczk -
Is it possible to get the precisely accurate playing position in milliseconds? I did find some title formattings that are capable to do it in seconds, but not milliseconds. I didn't manage to find a solution to this. If there's a solution, please remind me.
It's rather an important issue to consider when doing lyric stuff. Using title formatting would be great in my solution, so that I can fetch it through HTTP requests thanks to foo_httpcontrol, or any related APIs available when building components are welcome.
Last post by Sajadi -
For sure different sound formats have at least to some degree a "distinct special sound" - which results in many cases from one thing... progress of technology as time moves forward and formats get more efficient and better in general.
But quality wise Opus is better at lower bitrates and size wise often more efficient (compared with various encoded files in OGG/Vorbis and Opus - Opus encodings of the music which i encode are often smaller because Opus demands less Bitrate for the same quality setting) and either equal or marginally better (keeping more sound elements so the resulting converted songs feel more "full") at higher bitrates. Granted, the higher the bitrate the harder the differences can be detected.
Last post by kode54 -
The VPNs probably shouldn't be banned, but instead, the abusive users reported to the VPN provider. I suppose it also depends on the VPN provider, whether they care at all that people are using their services to evade bans or post spam.
Last post by EpicForever -
Guys, please. Let's leave "how to deal with jerks philisophy discussion" and other off-topics.
Once I also started thread requesting exactly this option - additional "undo-redo" level when editing tags. I am also sometimes annoyed, that after multiple edits on multiple tracks selection, one wrong edit causes that I have to cancel everything and start over. I suggested that instead Ctrl-Z and Ctrl-Y keys there could be Undo and Redo buttons added in Properties Window - in case usage of shortcuts would somehow interfere with contents of current operation stack, so there could be separate one, controlled only with buttons and independent of the existing one. I would consider this as just feature proposal for another major release of foobar - like 1.5 or something.
Last post by magicgoose -
If there's no clipping then the louder version is effectively having +1 more bit depth. +6 dB is roughly 1 more bit if you're using linear PCM sampling. It's "useful" in a sense that theoretical S/N ratio is +6 dB more, but nobody will be able to hear the difference likely.
Also if you multiply an already integer wave by exactly 2 so that the lowest bit is always zero (and you aren't adding quantization distortion nor any new information), FLAC will see this and compress just as well.
Before downgrading, I had looked for the UVC option in Win10 and couldn't find it. Now I've installed again v1.4 and found the option in foobar200 preferences. Keys are working again, also after one PC reboot. UVC option is enabled. I'll be vigilant when keys stop working and then will disable UVC integration to see what happens.
Last post by magicgoose -
Just saying, when you make loud bass sound waves, "perfect" woofers aren't enough, you need to make sure that no objects in your home resonate with it. A lot of metal/plastic/glass/etc. objects may add "distortion" because they literally jump and hit other hard surfaces, and they need to be fixed in place or dampened somehow. So if you hear a distortion, check if it actually comes from the subwoofer or it's something jumping in your room.
Last post by magicgoose -
Maybe SoX resampler is not necessary if the SACD component does that with the similar precision. I can't confirm for sure if SoX is necessary here because I don't know about the resampling algorithms used in the SACD component.
And yes if your SACD component is set to already produce 44.1k data then of course no further resampling is needed because it's already 44.1. Resampling from 44.1 to 44.1 will _at best_ do nothing at all.
> as long as the sample rate for DSD64 is a multiple of 44.1 then it should be good
Actually good resamplers like SoX can resample to any rate without any sound quality problems.