___AI mastering services___
Might be useful even for enhancing quality of instrumentals after separation (or your own mixed music)
Be aware that at least some advanced mixing beforehand may cheat the content ID detection system, so your song won't be detected. If some label prevents you from uploading their stuff on YT by blocking it straight after uploading a regular file, you may get a copyright strike after some time of uploading mastered instrumental as they also use the search engine on YT too to find their tracks at certain periods.
If you don't find satisfying results with the services below, read that.
Paid
https://emastered.com/ (unlimited free preview, 150$ per year)
Preview is just mp3 320kbps @20kHz cutoff, which is claimed to have a watermark, but it cannot be heard or seen in Spek. The preview file can be downloaded by opening Developer Tools in browser, and playing preview, then in "media", the proper file should appear on the list (don't confuse it with original file), now open the proper link in the new tab and open options of the media player and simply click download.
It's the most advanced and better sounding service vs all free ones I tested (even if you have only access to mp3, but I also listened to max 24 bit WAVs on their site with a paid account). Also, it's one of those, which are potentially destructive if you apply wrong settings, but leaving everything in default state is a good starting point, and works decent for e.g. mixtures and even previously mastered music to some extent, at least which does not hit 0dB (but e.g. even -1dB, but it is claimed to work the best between -3dB and -6dB). Generally I recommend it. Worth trying out.
Note for paid users - be aware that preview files can be mp3 files as well. So what you hear during changing various parameters, is not exactly the same as final WAV output.
https://distrokid.com/mixea (99$ per year/first master for free)
“[vs LANDR, BandLab and eMastered] I experienced that Mixea mastered with a much stronger sound and brighter (in a good way, the trebles are very clear) than the others.”
https://www.masteringbox.com >
https://www.landr.com/ (now also plugin available)
https://masterchannel.ai (15/20$ per month, only free previews, also can convert stereo to multichannel audio)
https://ariamastering.com/en/Pricing (from 50$ per month or 9.90$ per master, mastering based on fully analog gear and robotic arm to make adjustments in real time)
https://glowmastering.com/ (8$ for 5 masters, unlimited for 80$, 3 for free)
VST plugins
iZotope Ozone Advanced 9 and up (paid)
Version Advanced has a new AI mastering feature which automatically detects parameters which can be manually adjusted after the process. It works pretty well, and repairs lots of problems with muddy mixes (especially with manual adjustments - don't be afraid to experiment - AI is never perfect).
Mastering Assistant built-in the recent versions of Logic Pro DAW (MacOS only)
It can give more natural results than Izotpe above
AI Master by Exonic UK (paid)
master_me (free)
It contains a decent mastering chain which adjusts settings for you automatically for the song which can be changed later, and also you can change target ilufs value manually. By default, it's -14 ilufs and can be too quiet for songs already mastered louder, and it can become destructive while set that way for some songs
Free online services (all below remarks apply when mastering AI separated instrumentals)
https://aimastering.com/ (redirects to https://bakuage.com/app/)
wav, mp3, mp4 accepted, output: wav 16-32, mp3 320kbps, 44 or 48kHz
You can optionally specify a reference audio file.
Tons of options but not a comfortable preview while tweaking them. You can optionally specify the reference audio, uploading a file. Also, there’s one completely automatic option. Generally it can be destructive to the sound, even using the most automatic setting - attenuation of bass, exaggerating of higher tones.
Preferred options while working with a bit muffled snare in the mix of 500m1 model for instrumental rap separation result
(automatic (easy master) is (only) good for mixtures [vocal+instr]):
- True Peak, Oversampling 2x, AM Level 0.3, WAV 32. SAO, 0/22000 (the rest untouched)
For still too muffled sound (e.g. when lost in lots of hi-hats):
- YouTube Loudness, OVS to 1x and AM Level 0.2 and 24 bit (+ true peak, SAO, 0/22000)
Alternative (good for mixtures and previously mastered music with a bit muddy snare):
- YouTube Loudness, Target Loudness -8, Ceiling -0.2, OVS to 2x, True Peak and AM Level 0.3 and 32 bit, SAO, 0/22000
The most complicated tool, but the most capable amongst all free ones mentioned here so far. After two first files, it gets you into a short queue. Processing takes 2-3 minutes. Cannot upload more tracks than one at the same time. Great metrics, e.g. one measuring overall "professionality" of the result master. At this point, it can also start exaggerating vocal leftovers from the separation process. Equalize Loudness doesn’t do anything when checked just before download (probably only after when you click remaster).
They also have offline app: https://github.com/ai-mastering/phaselimiter-gui/releases/
with some features used on Bakuage/aimastering.com
“but most of the settings you want are on their site, their offline version is set and forget. (...) doesn't give you some specific settings to adjust.”
16-32 bit WAV output (now WAV is only in premium), any input formats. They have bad separation tools, but great, neutral mastering AI. It works very good for vinyl rips. You can get more than 5 tracks per month for free (don’t know how many - the 5 tracks limit is for separation, not for mastering feature, at least 30 worked in 2022).
The mastering feature is only available in the web version, so if you’re on the phone, run the site in PC mode.
24 bit -9 iLUFS / or without limiter does the best job in most cases for e.g. GSEP (the latter is when you don’t want to smooth out the sound). -8 tends to harm the dynamics of songs, but in some cases it might be useful to get your snare louder.
The interface has a bug when you need to pick your file to upload twice, otherwise you won’t be able to change parameters and confirm the upload process (also on mobile parameters not always appear immediately after you pick your file/pasted link enlisting the options manually doesn’t let you confirm the step to proceed to upload, and you need to retry picking the file, and now you can proceed).
Sometimes uploading is stuck for very, very long on 99% and if you leave your phone in sleep mode and return after 15 minutes, it will start some upload again on this 99%, but eventually it will return the error. You simply need to retry uploading the file (it will also stack at 99%, but it will still upload at that time).
Also, importing the same file via GDrive may not work.
Additionally, if you pick 32 bit output quality, when mastering is done, when you will want to download the file, in WAV it will show 24 bit, but the file will be 32 bit as you selected before.
It’s the most neutral in sound in comparison to the two below.
If you plan to master your own music, read “Preparing your tracks” here: https://moises.ai/blog/how-to-master-a-song-home/ I think these tips are pretty universal for all of these services.
Four presets with live preview, only 16 bit WAV for free, only WAV as input accepted (for the best quality convert any mp3’s to WAV 32-bit float (you can use Foobar2000), 64 bit WAV input unsupported).
If you see "upload failed", register and activate a new account in incognito mode and everything using VPN (probably a block for ISP which I had).
Judging by only 16 bit output quality (which is unfair comparison to 24 bit on moises.ai) and for GSep 320kbps files, I found it worse, and even the London smooth preset is not so neutral like moises in overall, and it can be destructive to the sound quality. But, if you need to get something extra from the mix if it’s blurry, that’s a good choice (while some people can find emastered too pricy).
BandLab Assistant mastering
First, you need to download their assistant here:
https://www.bandlab.com/products/desktop/assistant
Then insert the file, pick preset, listen, and then it is uploaded for further processing, and you’re redirected to the download page.
They write more about it below:
https://www.bandlab.com/mastering
Four presets - CD, enhance, bass boost, max 16 bit WAV output only. In comparison to paid emastered, it’s average. But in some cases it’s better than free mastering.studio when you have a muffled snare in the instrumental. On GSEP only CD preset was usable. The sound is more crusty than even LA Punch - more saturated (less neutral) a bit too bassy and compressed, but it may work in some songs where you don’t have a better choice and all above failed.
If your file doesn’t start uploading (hangs on “Preparing Master”), make sure you don't have “Set as a metered connection” option enabled in W10/11. If yes, disable it, and restart the assistant.
Straight after your file is done uploading, it is being processed, so don’t bother going to BandLab site too fast - sometimes it’s being processed even after download button appeared, where you start waiting in a queue even few minutes after you press the WAV button later, and you will not make this any faster.
On the side. The audio you hear during preview is not exactly the same as in result downloaded from the site. Preview is a bit louder, and stresses vocal residues more, and snare is less present in the mix, although the file is more clear, sadly it’s also 16 bit, in overall it doesn’t seem to be better. Also, the file doesn’t seem to be stored locally anywhere. But if you’re desperate enough to get this preview, fasten your seatbelt. If you processed more files before, close the assistant, and open again, now process the file, so preview can be played, pause it.
On Windows go to task manager, go to details, sort by CPU, RBM on BandLab Assistant.exe (the one with the most memory occupied)>Create dump file. Open it in HXD (located in temp), write in bytes per row instead of “16”, “4000”, find string “RIFF,”. If you cannot find it, it’s wrong process - make a dump of another assistant one (one of three most intensive). If you found the “RIFF,” delete everything above it (mark everything dragging the mouse to the top, with page up pressed and then keep shift pressed and left arrow to mark also the first row, then press delete), then save it as wav. The file can be played, but it’s too big. To find the end, go to “find” (CTRL+F), hex and write FF 00 00 02 00 01 00, find (it shouldn’t be at the beginning of the file - press F3 even more than once if necessary), mark everything dragging the mouse to the top with page up pressed and press copy (CTRL+C) and paste it into new file and save as wav.
You can also use Matchering. It works in a way that you provide a reference file, and it tries to match the sound of your audio to the reference you provided.
Reference file(s) to use
“Mastering The Mix” (all-in-one collection of reference songs in one file):
https://drive.google.com/file/d/1kqPmcVC3qvh_Mqd9vIssGUKpz3jTddPc/view?usp=sharing
You need 7zip with WavPack plugin to extract it.
Brown/Pink noise:
https://drive.google.com/file/d/1wJHKRb2SIgJZIc-J8kEDD1k4OQj_OXzp/view?usp=sharing
“Try to use this as reference track in Matchering to get nice wide stereo and corrected high frequencies.” zcooger
But you can use a whole song of your choice, or its short fragment (e.g. instrumental part to get better result of separation)
- New Colab:
- Old Colab:
https://discord.com/channels/708579735583588363/814405660325969942/842132388217618442
- UVR5 (in Audio Tools) - incorporates Matchering 2
- Songmastr can be used online instead of Colab, uses Matchering 2 (7 free masters per week).
or:
- https://linuxcreative.com/articles/matchering/
(at the bottom; hover the cursor around the two circles)
Although they don’t seem to allow changing output quality like UVR.
Be aware that there’s a length limit in at least UVR5, and it’s 14:44 (or possibly just 15 minutes). Instead of hit or miss by lots of reference files in one, you can also use simply one song you think will fit the most for your track. You can even further split it to a smaller fragment with e.g. lossless-cut in order to avoid reencoding. It can work even more efficiently that way.
Sometimes I use Matchering for different master versions of the same song when I have a few masters I like certain things in them, but none good enough on their own.
Usually, in the target file should be placed the file with the richest spectrum (but feel free to experiment).
Can be a target file e.g. after a lot of spectral restoration, which e.g. lost some warm and fidelity, and you need something from the previous master version.
You can also try to reprocess your result up to even 6 times, inputting a new file in target or reference each time, till you’ll find the best result. But usually 2-3 should do the trick, sometimes while using target and reference interchangeably for different result files.
For using Matchering in UVR5, necessarily check the option “Settings Test Mode” in additional settings. It will add a 10 digits number to each result, preventing you from overwriting your old files during multiple experiments conducted on your files. UVR doesn’t ask before overwriting!
Feel free to experiment with WAV output quality. Probably the further you’ll go from 24 bit, the more different your result will be after converting back to 16 bit by some lossy codec like Opus on YT. But if you care mostly about the result file, then simply be aware that you can use output quality to your advantage, knowing in what way specific bit depth affects output results. E.g. the muddier results start with PCM_32 (non-float), 64 bit has it too, but additionally with some grittiness, but you can convert it back to 16 bit without dithering e.g. in F2K (not in DAW to avoid another conversion to 32 bit) to glue it back, when having more clarity than PCM_32. Sometimes 16 bit can be good to glue well sounding audio together with loudly sounding snares already, but can be muddy frequently or harsher than e.g. 32 bit non-float. Usually your result will be not so good in most cases, hence I’d encourage using higher bit depths than 16 bit here, but 24 bit can make your audio too bright at times, hence in such cases you can check 32 bit float and non-float. There’s no simple setting working for every song, but the most universal setting I found so far is using non-float 32 bit and sometimes convert it to 16-bit manually or 64 bit converted to 16 bit. These are the most balanced settings across the whole song.
Sometimes it can be good to have the richest file on a spectrogram as a target file, as it won’t be lost after processing.
Matchering can be generally useful when you have different versions of your masters, and you’re running in circles finding the best one. Then you can use such different versions as target and reference (or in reverse), check what sounds the best, get the result, use in one of the fields, retry, and the same up to 4 times till it sounds the best. Then you could potentially master it further and/or separate into stems and bring the session back from this place.
If you need more customizable settings for Matchering, e.g. controlling limiter intensity, or disabling it completely, consider using ComfyUI-Matchering (standalone/portable ComfyUI package for CPU or Nvidia: new_ComfyUI_windows_portable_nvidia_cu124_or_cpu)
- .masterknecht
https://masterknecht.klangknecht.com/
Web-based competitor of Matchering (it’s not associated with Matchering). All the processing is done locally on your machine without uploading files to a server.
The results using default settings usually sound a bit softer/warmer to those from Matchering, output is 48kHz, plus there’s much more customizable settings.
Some exemplary experimental settings for more clarity (32-bit float):
- resampler linear, limiter disabled
- resampler linear, limiter disabled, dither RPDF
- resampler linear, limiter peak
- resampler linear, dither RPDF (the brightest suggestion)
Once you change the settings click process again (no need to choice the files again).
Others
Windows app
Some newer AI:
https://huggingface.co/spaces/nateraw/deepafx-st
Also try this one:
https://github.com/jhtonykoo/music_mixing_style_transfer
https://huggingface.co/spaces/broadfield/music_mixing_style_transfer
https://github.com/rupeshs/neuralsongstyle/tree/master
(probably older)
Or:
https://github.com/joaomauricio5/AssistedSpectralRebalancePlugin
Others (by bizzare808)
https://reedmayhew-audiomaster-ai.hf.space/
https://entrepeneur4lyf.github.io/Web-Audio-Mastering/
https://www.munute.com/ai-mastering
ChatGPT
It can now master songs based on prompts and whatever you ask to make it sounds like. In the video below, the author wanted to make three instrumentals sound like Juice World. One result was decent, in the other one there was an issue with ovecrompressing/overlimiting, so the guitar was fading in/out once some other instrument was kicking in. Some prompts might fail to give you the result file, and he provided the examples.
https://www.youtube.com/watch?v=0kGJVgiyhAk
“GPT does a lot of things right now, but the biggest problem is that it can't get larger files (wav) into the buffer and thus can't process them. Compared to unstable work results, not working is more serious.” - tat_evop1us
https://beatstorapon.com/ai-mastering (only mp3 192kbps for free)
For enhancing 4 stems separations from Demucs/GSEP:
https://github.com/interactiveaudiolab/MSG (16kHz cutoff)
(Windows/Mac paid software)
"corrects pitch, improves volume and makes every file ready to play anywhere (...) add warm" and dynamics, remove clipping.
Mastering services I'm yet to test:
Landr, Aria, SoundCloud, Master Channel, Instant Mastering (iirc April fools joke), Bakuage, Mixea.
AI mixing services
AI online auto-mixing service. Various instruments, genre settings, stem priority, pan priority.
1 free mix per month.
Might be useful for enhancing 4 stem separations.
"I tried 2 songs with it. Wasn't really pleased with results"
"The biggest problem I had [...] while I am trying to balance my vocals in instrumental like Hollywood style"
Other tool by Sony (open-source)
https://github.com/sony/fxnorm-automix
You can also train your own models using wet music data.
A new free tool by Sony:
https://github.com/SonyResearch/MEGAMI
https://www.arxiv.org/abs/2511.08040
HyMPS list of AI/CML tools
AI mixing plugins
iZotope Nectar
iZotope Neutron (Mix Assistant)
Sonible Pure Bundle
Creating mashups and also DJ sets (two options)
It can give better results than manual mixes performed by some less experienced users (but I doubt it will work with more than 2 stems).
iOS only app (currently for US region only)
"Ripple seems to be SpongeBand just translated into English, it was released last year: https://pandaily.com/bytedance-launches-music-creation-tool-sponge-band/
(more info about its capabilities)
Back then, it only didn't have separation to 4 stems (but now the separation feature is defunct, anyway).
_______
For enhancing vocal track you can use WSRGLOW, and better yet, process it through Izotope RX (7-9) spectral recovery tool (in RX 10 it’s only in more expensive version irc), and then master it, or send it somewhere else above.
https://replicate.com/lucataco/wsrglow
There are a lot of requests for music upscaling on our Discord. You can use online mastering services as well. Technically it's not upscaling in most cases, but the result can be satisfactory at times.
If you try out all solutions, and learn how they work and sound, you can easily get any track in better quality in few minutes.
For very low resolution music (if you manage to run it):
AudioSR - used more often then the below, lately (voc/inst)
Audio Super Resolution
https://github.com/olvrhhn/audio_super_resolution
hifi-gan-bwe
https://github.com/brentspell/hifi-gan-bwe/
More details and links, Colabs for these in the upscaler’s full list
If you want to start making your own remasters (even if your file is in terrible quality, especially 22kHz):
https://docs.google.com/document/d/1GLWvwNG5Ity2OpTe_HARHQxgwYuoosseYcxpzVjL_wY/edit?usp=drivesdk
Might be useful also for low quality, crusty vocals, but it is also a guide for mixing music in overall but focused on audio restoration as well.
___Best quality on YouTube for your audio uploads____
- If you already have a ready video which is not just a one frame (e.g. a cover all over the video), download MKVToolnix and replace audio track with lossless one instead of rendered lossy track. You will avoid recompression or reencoding, unlike it is during rendering normal video.
- If you can, upscale the video to at least 1440p or greater. It will avoid deferred transitioning of your AAC (16kHz) audio stream to Opus (20kHz) when your video gets popular, or it's old enough (for current YT audio format, check statistics for nerds). QHD/+ makes your video play in better Opus codec from the beginning, and it will sound better than after deferred transition from AAC to Opus on FHD clip (Opus audio streams checksums differs in FHD and QHD videos despite the same video source file and most likely something is broken on YT side during the process, though both Opus files are 20kHz, so the file in FHD is not recompressed from AAC, perhaps from other audio file created during YT rendering, but not from the source video).
- Alternative - if you have just one image to make a video of it (e.g. cover), make sure it’s at least 1440p or greater. If not, simply upscale it (e.g. XnView has some basic upscaling filters). Then place the image nearby this batch FFmpeg script with your lossless audio files. It will render videos with the same audio streams like original files, but muxed into your output MKV files (you can check in Foobar2000 for Audio MD5 comparison or by using AudioMD5Checker, if MD5 checksum is not embedded when looking in F2K file properties) so it won’t be recompressed on your end while making a video for upload on YT (yes, YT supports MKV!). It’s faster than MKVToolnix and you can convert multiple files with the same image at the same time (it's very fast, incomparable to normal video rendering, and output is only 1 FPS, so it will buffer in YT also very fast).
- You don’t have to wait till YT stops processing your HD version for Opus to appear. It happens at a point when FHD resolution appears before QHD when processing is still in progress. So check it out from time to time before you hit the publish button.
- Because Opus is 16 bit, and your input audio file in Matroska container might have higher bit depth, it’s good to compress your input file to Opus VBR 128kbps for testing purposes to check how it will sound on YT (of course don’t use it later for MKV file). Downsampling performed by the encoder can occasionally introduce some unwanted changes to the sound. It’s the most noticeable when audio input is 64 bit, but smaller can be still good enough.
- YouTube videos from early 2010 on archive.org have 192kbps AAC for 1080p (example) (thx theamogusguy)
- If you deal with some harshness on your YT audio uploads with original 44kHz sample rate audio in the uploaded video file, consider upsampling them manually to 48kHz before upload. It will bypass the built-in Opus resampler. You can do that using e.g. Izotope RX (smooth). Although it can sound smoother just because of forced upsampling on file import to 32-bit in RX Editor and downsampling during export, esp. without dithering.
Alternatively, you can use dBpoweramp/SSRC (F2K plugin) or SoX in the newest Foobar2000 x64 and save the output as lossless format.
Currently the best resampler on the Hydrogenaudio SRC chart is:
https://github.com/rorgoroth/mingw-cmake-env/releases/tag/latest
Usage:
ffmpeg -i "C:\input96or48.wav" -af asf2sf=dblp,ardftsrc=44100:quality=61656210:bandwidth=0.9941249 -c:a pcm_f32le C:\output44.wav
If you don’t deal with clipping, delete -c:a pcm_f32le to use just 16-bit output (other formats).
24GB RAM recommended. It doesn’t track progress. It can take more than 10 minutes in an insufficient RAM scenario (and normally less than 5 for 5-minute audio).
PS. E.g. 96>44kHz resampling on 4:20 file requires 40.5GB of memory. With 16GB of physical memory, it will require 25GB of SWAP (by default, it will be used from C:\), and if you run out of disk space, it will briefly show "Cannot allocate memory" during executing the command. Make sure that you didn’t limit the SWAP manually. Consider turning on SWAP on two SSDs to accelerate the process.
- Since now MVSEP supports batch API conversions, you can use Case Changing in Ant Renamer to reestablish uppercase letters to song titles for your YouTube uploads.
So if you want to batch upload on YouTube the name will not appear as "song artist song title" but "Song Artist Song Title" (since YT removes dashes and commas etc.)
- It seems like 720p is now enough to get Opus after upload (thx dca100fb8; points 9 and 8)
- Safari browser probably still don’t support Opus, so it rather always uses AAC 128 kbit/s 44kHz audio stream on YT for all videos instead (should be available to check in PPM on played video>Stats for nerds)
Note: Google can block your account if you use AdBlocks on YouTube. Use some spare one you won't regret losing for it.