8/5/2023 0 Comments Magix music maker mp3 encoder![]() Sorry to dig up this thread again, but did this ever get resolved? If you need to archive or save in a smaller format in between working on a given sound file, use the truly lossless FLAC (or Apple Lossless in the Mac/iOS world if you prefer). Passing an already lossy file through more lossy encoding is often called 'transcoding' and it is a sure prescription for gnarled, mangled high end and more - and it will, indeed, tend to make you think of really crappy, low-bitrate lossy encoding. Always use uncompressed wav or AIFF files for processing, editing, etc.) ![]() One last - but maybe crucial point - never, ever, ever use an mp3, AAC, Ogg, or other lossy-compressed sound file for MORE processing and then re-encode it! (If you can possibly avoid it.) ONLY use such files for playback. ) The 'perceptual encoding' algorithms on the second pass will go back over the already encoded file (You typically have to use a 'wrapper' or command line switches in 'terminal' mode. Me, I always use LAME's slow/high-quality setting because it's just not that much slower and sound quality matters more to me than one-time processing waits. It's also important to remember that all encoders (we're more or less down to the 'official' Fraunhofer and the open source LAME) have different modes of operation, most importantly for this conversation, a tradeoff between speed of processing versus quality of processing. But because of licensing issues, most mainstream commercial software that includes mp3 encoding uses Fraunhofer.) (I'm among the substantial cohort who tend to like the LAME encoder better than Fraunhofer. So I don't know what's in the current version, but I would suspect it's Fraunhofer - particularly now that the Sonic Foundry software that used to be owned by Sony is now owned by German soft-boxer, Magix. My version of Soundforge is old and only came with a 'trial' version of the Fraunhofer codec, good for a limited number of encodings. Of course, properly made mp3s tend to sound better the more sound data we leave in them - but not all codecs are created equally, even in the 'mp3' arena. :(įirst, the term bitrate is a measure of size per time, nothing more. I'm sick at the thought of how many of my songs might be ruined and a lot of them I won't have unaltered backups for. I really like Sound Forge and hate to have to start using two editors or going with something else totally. How on earth is Sound Forge saving my mp3s at the wrong bitrate, yet having the same file size? And if it's the same filesize, how is more frequency information missing that what should be? In most cases, the file sizes are the same no matter what editor I use. Every single one correctly re saves or converts to mp3 and the spectrum analyzer shows a much greater amount of frequency range in the mp3. ![]() So I began to download various freeware editors and even a video editor. When I re save an mp3 in either version of Sound Forge, my spectrum analyzer says that it is in fact 320kbps but the frequency information reads more like it is 192kbps. So just to make sure you guys understand what I'm saying. ![]() I began to notice that when I looked at mp3s that were created by Sound Forge, the spectrum analyzer would show that my mp3s were identified as 320kbps yet the frequency graph shows that they are a much lower bit rate! This is beyond bad because it is possible that I have messed up tons of mp3s and songs that I edited and resaved. Sometimes I will just give older songs a volume boost with the SF limiter. Sometimes the source is WAV and sometimes it is mp3. I will often use SF and FL Studio to make my own remixes or edits of songs. Recently I started using a spectrum analyzer to look at the mp3s that I DJ with. I have SF Audio Studio 10 and SF Pro 10.Īnd recently I discovered something very troubling and I'm pretty unhappy about it because I think I'm going to have a lot of further disappointment ahead of me. So I have used Sound Forge for at least a decade now.
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