mastering for itunes plugin in wavelab

Hi,

I have been reading about “mastering for itunes”, and as I understand it, you want to use a plugin that lets you control if the file you are playing will be clipped when its later converted to AAC format. He is using the plugin in this video: (4:52 into the video)

https://www.youtube.com/watch?v=r8HTmf--Wgs

But he also says in that video that it only works for mac. Do you know if there is any plugin that I can use in wavelab on my pc for this?

First, let’s not confusing “mastering for iTunes” with the official program Apple calls “Mastered For iTunes”. At this time, you have to be an approved mastering engineer/studio by Apple in order to submit official “Mastered For iTunes” masters. However, if you’re not on this list, some of the concepts can still apply and be done when you master for iTunes and other streaming/digital stores.

With “Mastered For iTunes”, the main goal is to not have any clipping or overs when your master WAV files are encoded to AAC for the iTunes store. The other benefit is that you can submit 24-bit/high sample rate WAV files for the AAC (end user files) to be encoded from which sound arguably better than 16-bit/44.1k.

The first step about the audio not clipping after the lossy encoding is something you can and should consider for not just iTunes releases but all digital releases. Some mastering engineers leave up to 1dB of headroom to make sure the myriad of encoders the audio will face do not cause the peak levels to exceed 0dBFS and potentially have bad artifacts.

The large majority of digital distributors still ask for 16-bit/44.1k WAV files so all this applies beyond just iTunes releases. You can also still do a standard iTunes release which requires 16-bit/44.1k WAV but you can still make sure there is enough headroom for the AAC encoder to not cause peak levels over 0dBFS.

The second step as mentioned is special to Mastered For iTunes at this time but websites such as Bandcamp and SoundCloud also accept 24-bit/high sample rate files but beware SoundCloud’s port 128kbps audio stream which is sure to induce more clipping and artifacts than something that does 320kbps mp3.

So, I think it’s good practice to be mindful of what happens after lossy encoding no matter what platform you master for because these days, it’s inevitable that the music will be data compressed using a lossy encoder somewhere down the line. Even a CD user these days is likely to just import the CD tracks into their iTunes or media library as a lossy format without even listening to the actual CD.

Anyway, back to your question. WaveLab comes with a similar plugin to the one that Ian Shepherd is using in the video called Encoder Checker. I honestly think the one that Apple provides free is garbage and I don’t use it. Check for a plugin in the master section or Playback Processing slots called Encoder Checker.

Also, one I highly recommend is Sonnox ProCodec (https://www.sonnox.com/plugin/fraunhofer-pro-codec) which I believe is on sale for one more day for their November sale. Aside from what I’ll describe below, it’s a great tool for checking editing/metadata though WaveLab can usually do 99% of the metadata work for you.

What I like about Sonnox ProCodec is that you can analyze the files both in real-time and offline. The offline analysis in my opinion is more useful because you can get repeatable results regarding the peak level changes after encoding. When using these tools on live playback, you get an idea of the sound of the encoder but the peak levels are not 100% repeatable because the bitstream of the codec and the start of the eventual rendered WAV of each track are not in sync. So on live playback, you will see slightly different results with the peak levels. This is discussed in the Sonnox manual.

Unfortunately, at this time there isn’t a codec that is 100% matching the “Mastered For iTunes” (known as iTunes+) available on Windows so that is why Ian says you need to be on a Mac, but Sonnox states that “On Windows computers, the closest approximation to the iTunes Plus codec is the Fraunhofer AAC-LC codec set to highest quality VBR at 256 kbps.”

Thanks very much Justin I learn so much from you! I get back with some questions after I have read your answer more in detail!

No problem. I meant to say that I’m not familiar with the built in Encoder Checker on Windows specifically and if it offers an AAC option but Encoder Checker is the same concept as the plugin shown in Ian’s video.

I am a little confused about how many formats I shall render to when I have recorded an album. I want to do an audio cd, and send a digital version to cdbaby, which put the songs up on many other sites spotify, itunes etc…, I am also thinking about doing a vinyl too =)

But would you suggest me to do this versions:

  1. A physical audio cd
  2. DDP
  3. Digital 16 bit 44 khz wav files (with its peak at -1 db) (for cdbaby)
  4. 32 bit floating point files (to have as archive, if I want to get back to the project sometime?)
  5. 24 bit files (if I want to do a vinyl?)
  6. 16 bit files with its peak values at -0.3 (same as the cd)

Would it be a good idea to render all this versions, or can I skip some?
Do I need any versions beside those? Maybe aax, or mp3 and if so what are those used for?

There’s no “right way” to do it but I can offer my take:

I first deliver the master as DDP using HOFA DDP Player Maker to easily send my clients a DDP with easy to use DDP Player to approve the master. This allows them to be sure of all the titles, spacing between songs, and the mastering itself.

I set the final limiter ceiling somewhere between -1dB and -0.5dBFS deepening on the material. I’m also using a limiter that is good at detecting and preventing intersample peaks so sometimes -0.5dBFS is safe for most future encodes.

Whatever ends up happening, when the DDP is approved I then render the 16-bit/44.1k WAVs for the client with the same settings.

From there, it depends what the project needs. I have a form on my website that lets clients choose what they need. This often includes:

-24-bit/high sample rate WAVs within the Mastered For iTunes Spec as well as Bandcamp, SoundCloud, and possible near future digital distribution.

-Vinyl Pre-Master which is usually a 24-bit/high sample rate WAV of each side of the record, with some special adjustments to translate well to vinyl, no digital limiting etc.

-Cassette Pre-Master which is normally 16-bit/44.1k, same processing as the digital master but I render one file per program (tape side).

WaveLab CD Track Groups make it very easy to render a WAV for each vinyl/cassette side and then make a PDF report with the marker times and other info.

-Reference mp3 and/or AAC files because even though clients can make mp3s on their own, I prefer to make them myself because I can encode from the undithered 32-bit floating point stream, and tweak the limiter ceiling to not cause overs. Also, thanks to WaveLab I can easily tag the mp3/AAC files with all the metadata and artwork that clients like to see to make it look like a professional release. WaveLab makes it easy to transpose the CD-Text info and other info from the montage over to ID3 metadata. These never get used for most professional distribution but more and more people want an mp3 or AAC version to share directly with others, and some download card services require mp3 files because they don’t do any file conversion. They just host the files you give them for others to download directly.

I never make a physical CD version unless it’s needed in the moment for CD duplication but most places accept DDP. CD-R doesn’t have a great shelf-life and is not a great format to start with in my opinion.

I also never render 32-bit floating point because with my workflow, I have easy access to the mostly mastered stuff (post analog gear capture) but without the final digital limiting which if I wanted to revisit years and years later and WaveLab or the plugin no longer works, I’m sure I’d want to redo that step anyway. I of course keep all the rendered masters though for archive. Making a 32-bit float version wouldn’t be a bad idea through, I just know for me it might not be used. I would start over with the post analog capture version and go from there.

MrPicker, you already linked to an Ian Shepherd video, but a good tip I think for you would be to listen to the latest edition of Ian Shepherd’s podcast ‘The Mastering Show’ (if you didn’t already know it). It’s exactly about mastering for streaming platforms: http://themasteringshow.com/

Whatever ends up happening, when the DDP is approved I then render the 16-bit/44.1k WAVs for the client with the same settings.

That 16 bit file, is that for digital distribution or what is the purpose of that? Or is it only the 24 bit file you mention later that is for digital distibution? Or both?

-24-bit/high sample rate WAVs within the Mastered For iTunes Spec as well as Bandcamp, SoundCloud, and possible near future digital distribution.

Do you change the limiter peak for this 24 bit file, or do you keep it somewhere between -1 and -0.5dbFS?

-Vinyl Pre-Master which is usually a 24-bit/high sample rate WAV of each side of the record, with some special adjustments to translate well to vinyl, no digital limiting etc.

No digital limiting? Shouldn’t I use the maximizer at all (which I use as limiter) in wavelab at all for the vinyl version?
If not, should I only raise the volume as much as possible before it clips, without any kind of limiting?

I also never render 32-bit floating point because with my workflow, I have easy access to the mostly mastered stuff (post analog gear capture) but without the final digital limiting which if I wanted to revisit years and years later and WaveLab or the plugin no longer works, I’m sure I’d want to redo that step anyway. I of course keep all the rendered masters though for archive. Making a 32-bit float version wouldn’t be a bad idea through, I just know for me it might not be used. I would start over with the post analog capture version and go from there.

Excuse me, but what exactly is “post analog capture”? Is that the files as they are when exported from cubase, before imported to wavelab for mastering? (I use limiting with the maximizer in wavelab, and also do some eq, but you mean its a good idea to save the 32 bit version before using the maximizer?)

  1. The 16-bit/44.1k WAV of each song is for basic digital distribution at this time (Spotify, iTunes, Pandora, Amazon etc.). It is rendered from the same montage as the approved DDP so it is identical to what the client has approved. If the client is making CDs, then we also have an approved CD replication ready DDP. If not, we still had a good way of approving everything, and clients can burn their own CD-R which some still do, more than I thought.

  2. This version also has the limiter output ceiling close to or at -1.0dB because aside from lossy encoders, this audio is likely to have a sample rate change in the cases of Mastered For iTunes or if Bandcamp creates mp3 from the 24-bit/high sample rate WAV masters. So between the sample rate conversion and the lossy encode, there are two places where peak levels will increase. I don’t use the built in Resampler of WaveLab, I use a 3rd party SRC so this involves making “Custom Montage Copies” to recreate a montage at another sample rate and keep all the markers, plugins etc. identical.

  3. There is no rule for vinyl but it’s widely believed and confirmed by some lacquer cutting engineers that loud digital masters that have limiting applied do not translate well to vinyl, and can result in a quieter and less exciting piece of vinyl. Again, no rules, only trial and error. The limiter can also affect the balance of instruments if it’s heavily used on the digital masters so it’s a delicate practice to remove the limiter for the vinyl, you don’t want to change too many things. It’s an art.

  4. When I say “post analog capture” I mean these are partially mastered files that have been processed to some degree with plugins, and then through my analog mastering gear, and then captured again to digital. They often also have some clean up work with RX5 for noise, clicks, pops, and other things. From there I bring them into WaveLab for finalizing which includes sequencing, final limiter/dither, and maybe some small tweaks per song if needed. Then of course all the stuff WaveLab is great at such as track IDs, metadata, DDP, etc. From here, with some slight adjutants I can render all the various master formats needed. I usually have a few different montages with the slight tweaks for each project so the settings are 100% recallable. I of course only make all the formats after the initial DDP is approved as to not waste time.

The things I normally save after a project is done are:

-original un-touched mixes
-all the session work of going to and from the analog gear
-the “post analog captures” and WaveLab montages
-all the master files I render and send to clients

You’d be amazed how many people email me weeks, months, or years later looking for their master files that they failed to save or store safely. Luckily I keep it all so when they ask, I can deliver.

Hello
I´m using WaveLab Maximizer to master my tracks, set it to -1,5dB and then Brickwall Limiter, set at -1,0dB; as I master to upload to Soundcloud. The question I want to ask is how can I control the dinamic range I get? through the mix knob?