In this tutorial, I’m going to run through how to make a self-master of your audio. I’ll be using Ableton Live, but similar principles can be applied in any digital audio workstation (DAW) software such as Logic, Cubase, Pro Tools, Reaper, FL Studio and so on. The tutorial assumes you’re working in a DAW of some kind.
Mastering is the process of finalising audio and preparing it for its intended use, e.g. being cut to vinyl, distributed by streaming services, put onto CDs or played on the radio. Typically this process is done by skilled professionals with specialised equipment, who will charge a fee for mastering tracks.
To be clear, I wouldn’t suggest that self-mastering is a substitute for professional mastering in the majority of cases. An experienced mastering engineer will be able to do the job to a high standard, and won’t have the inherent biases you bring to your own productions.
Nevertheless, self-mastering definitely has its place in the era of digital music production. It’s good for situations where you want something close to a mastered version, but it wouldn’t make sense to get a full pro master. For example:
- If you want to try out some of your own productions in a DJ set, it can help to have a version of your track where the loudness is comparable to released music. If you just bounce your mix with nothing on the master bus, your track will probably sound quiet by comparison to mastered music, at least in modern styles such as dance music, pop etc.
- If you’re sending out demos and you want them at a level of loudness that is comparable to finished tracks (I’ve seen some labels say they prefer to receive self mastered demos, and others say they don’t want any mastering on a demo, so this varies.)
- If you have audio that you want to sound good, but where pro mastering would be excessive, e.g. for uploading audio clips to social media, running a DIY podcast, releasing your own music on a personal Bandcamp page, or small projects where the budget won’t stretch to mastering.
I’ve also found that self-mastering seems to help my mixes. It gives a rough approximation of mastering, and this often reveals mix problems. I can then go back into the mix and fix it.
With modern digital audio software, it’s possible to make a self-master that does the job without too much difficulty.
What is involved in making a self master?
As a general principle, it makes sense to do as little as possible because it’s very easy to ruin a track with too much processing. If you find you need to make major changes to get a track sounding OK, there might be problems that would be better fixed by going back to your mix.
There are programs such as Izotope Ozone and T-RackS that can work as a standalone solution for making a self-master. Ozone has a master assistant takes you through the process and simplifies it. Ozone’s Elements version is a low cost, reduced features version which runs as a plugin, and might be a good choice if you want the simplest and cheapest possible way to make self-masters.
I prefer to self-master with more manual control, by building up my own mastering chain, i.e. a set of plugins to perform the various functions required. The basic principles involved could also be used when working with Ozone.
The process I recommend is:
- 1. Make a pre-master of your mix
- 2. Listening, referencing, metering
- 3. Equalisation – making small adjustments
- 4. Limiting – to get your track to the required loudness
- 5. Output the self-master – in the formats you require
There are all kinds of other things that might be added into mastering, such as tape emulation, compression and saturation and mid-side processing. In this tutorial, I’m aiming for simplicity – the bare minimum required to do a basic self-master – so I won’t cover these optional extras.
Let’s go through these steps in order. I’ll be working on a short section of techno track of mine called Quantize. This track was released digitally with professional mastering, so we’ll be able to compare the self-master with the pro master at the end.
(The audio clips in this tutorial are 320 kbps MP3s. So they won’t give a completely accurate representation of the full quality uncompressed audio of the self-master, but should be close enough to give a sense of what’s going on.)
1. Make a pre-master
You can either set up self mastering on the master bus of the DAW project where you’ve created your mix, or you can set up a separate project to do the mastering. Either way, the mix should be peaking around -6dB. A little higher is fine, but definitely no more than -3dB. The reason for this low level is to avoid any kind of distortion, which in some cases can happen at levels below 0 dB.
For the purposes of this tutorial, we’ll use the second approach, mastering separately to mixing. So we’ll begin by bouncing down the mix as a 24 bit wave or AIFF file, aiming for it to be peaking around -6dB.
The resulting audio is called a pre-master, i.e. a finished mixdown ready for mastering.
In general it’s best to make the premaster at the sample rate at which you’ve recorded the track. For me that’s 44.1kHz.
It’s important to use 24 bit for a pre-master, as this allows for greater dynamic range than a 16 bit file.
Usually a pre-master should have no processing on the master bus, with all processing left to the mastering stage. Like all rules, this can be broken, but for this tutorial let’s keep things simple and bounce the pre-master with no processing.
The pre-master can then be dragged and dropped into a new project. Make sure the channel you’re using is set to stereo, that the sample rate of the project matches the sample rate of your pre-master, and that there are no additional options enabled that could mess up the audio.
As I’m using Ableton Live, I’ve double clicked on the audio file to open up the additional settings, and turned off the warp mode which changes the tempo of the audio – not what we want for mastering. I’ve also made sure high quality playback (HiQ) is turned on.
Here’s a clip of how the pre-master sounds at this point. You may notice it sounds quiet, which is because it’s peaking at -6dB to give me plenty of headroom for mastering.
2. Listening, referencing, metering
Before making any adjustments, listen carefully to your track, paying close attention to the whole frequency spectrum. Switch between monitors and headphones if you can, and shift the focus of your attention repeatedly between the low end, the mids and the high end, and then the whole track altogether. As well as listening for what frequencies might be lacking, try listening for what there is too much of, for what sticks out or disturbs the mix.
Listening to my premaster, subjectively the overall frequency balance of the mix sounds quite good to my ears. The kick thumps but doesn’t dominate, which is about the right balance for this type of atmospheric techno. In the mids and highs, the mix is a bit polite or slightly muffled, so it might benefit from a slight lift in this area. Listen to the pre-master yourself and see what you think.
The next step is to do some reference listening. This is crucial. As you listen to your own track repeatedly adjusting the mastering settings, you’ll lose perspective on what it needs. A reference helps re-calibrate your listening.
Select a few released tracks that you think are good examples of the sort of thing you’re aiming for. For my self-master, I’ve chosen one of my favourite recent ambient techno tracks, and downloaded the WAV file from Bandcamp so I have it at full quality.
Released tracks will be mastered and therefore will be much louder than your pre-master. To compare them for the purposes of EQing, you’ll need to reduce the playback volume of the reference track as compared with the pre-master, so that they sound at a similar loudness.
There are plugins you can use to help A-B compare your track with references, but I find it easy enough to just drag the reference into a separate track in the DAW, adjust the level so it sounds approximately as loud as the pre-master, and use mute / solo to compare.
In my case, comparing my track with the reference I’ve chosen reinforces my initial impression that my track might benefit from a slight boost in the high mids and maybe also a little high end sparkle.
At this point, it’s helpful to add a frequency spectrum analyser to the end of the mastering chain. Visual metering is no substitute for careful listening, but can be used to confirm or contradict what you’re hearing. Also, sometimes a spectrum analyser can help you spot particular problems that may not be readily audible, such as very low or very high frequencies, or resonant peaks that might need to be tamed. You can also run your reference track through a spectrum analyser and see how it compares to your track.
In Ableton Live, you can find a spectrum analyser plugin in audio effects > utility > spectrum.
Most DAWs will have some kind of frequency analyser. In Logic there is an excellent Multimeter plugin which does frequency spectrum analysis and various other useful things like loudness metering and checking stereo phase coherence.
Here’s a snapshot of how my track looks in Ableton’s spectrum analyser:
The big peak around 50Hz is the kick drum. The area from around 2kHz to 10kHz seems slightly recessed – I wouldn’t want to massively boost this area as it could make the track sound brittle or tinny, but it does confirm my sense that a gentle lift here might even things out. There also are a few strong peaks around the 1kHz area, which might benefit from a little cut – or at least, I’d want to avoid boosting that area too much.
Another important area to look at is the very low end, below 20-30Hz. These frequencies are infrasonic, so will be mostly inaudible, and they use up valuable headroom, so I tend to use a high pass filter to roll them off. In this track, you can see that the level around 20-30Hz is similar to or even higher than the level in the high mids and highs (2-10kHz), so this very low sub is definitely something we should try to reduce.
The same principle applies to frequencies above 16kHz or so, towards the top of the human hearing range. If you can see a lot of energy in this area in the spectrum analyser, then some filtering here might be sensible. In my track at the high end, the frequency plot drops off upwards of around 10kHz, so there’s nothing I can see that needs fixing here.
We’re now ready to add some equalisation (EQ) into the master chain.
Note that everything we’ll do here is informed by the analysis from the previous stage. That’s crucial – without a diagnosis of what the track needs, EQ is a bit like shooting in the dark, and can make things worse.
First I’ll add Ableton Live’s utility plugin to the channel, and use that to make the bass mono. Human hearing can’t process directionality at low frequencies, so there’s little sense in having any stereo width down there. The utility plugin has a handy ‘bass mono’ feature and the default setting of 120Hz (i.e. everything under 120Hz is made mono) is fine. You could raise this a bit if you like, to 150-200Hz.
Next I’m going to try using Ableton’s stock EQ8 plugin to address the frequency issues identified in the previous section. That might give a pro mastering engineer heart palpitations, because EQ8 isn’t really designed for mastering. But the point of this tutorial is to keep things as simple as possible, with the aim of creating a rough self-master that is ‘good enough’.
To my ears, EQ8 sounds excellent. It’s neutral, clean and transparent, which is helpful for mastering. It also has the flexibility to set each band as a filter or peaking / bell curve EQ, with precise control over the resonance or Q of the peaks (i.e. whether they are broad or tight).
If you’d prefer to use an EQ plugin that is more specifically tailored for mastering, there are lots of third party options. I’ve found Brainworx bx_digital to be very good, albeit quite complex. Ozone also includes EQ modules. Other options suited for mastering include FabFilter’s Pro-Q3, SIR StandardEQ, DMG EQuilibrium, and plugin versions of highly regarded ‘character’ mastering equalisers such as the Dangerous Music BAX EQ and Chandler Curve Bender. There are debates about the merits of linear phase versus minimum phase EQs for mastering, but again for the sake of simplicity I won’t get into that here.
Whichever plugins you use, some useful general principles for EQing when self-mastering are:
- Do as little as possible. If you need to make big adjustments, it probably means the mix needs some work.
- Try to cut rather than boost. Cuts sound more natural.
- Aim to make the curve of boosts wide (i.e. low Q value) for a subtle effect. Cuts can be a bit narrower if you need to target specific frequencies.
- Use parametric peak / bell curve EQ bands and high pass / low pass filters in preference to shelving filters. This might be a personal thing, but a shelf EQ can boost all frequencies above or below where it is set, potentially amplifying very low and very high frequencies outside the range of hearing, which then use up headroom. Shelving is a bit of a blunt tool for mastering; I find parametric peak EQs more defined.
- With a peak EQ, don’t cut or boost more than 2dB.
- With high pass or low pass filters, 12dB per octave is a good starting point (in EQ8 this is the default slope).
- Constantly A-B compare by switching the whole EQ in and out, the individual bands in and out, and by going back to compare with your reference tracks. To compare effectively you need to try to match the levels with and without EQ. For example, if your EQ is increasing the overall level of the track by 1dB, then you should reduce the master gain on the EQ by the same amount, so the comparison is fair.
After adding EQ8 into the track after the utility plugin, and with a little careful experimentation and listening to the results, here’s what I ended up with:
- A 12dB / octave high pass filter set to 25Hz to remove the very low infrasound. This is the slope you can see at the left hand side. The idea here is that we are cutting inaudible frequencies to free up a bit of headroom, so I adjusted the frequency until there was no noticeable audible change, listening on headphones with good low end extension. The difference should be inaudible but show up in the spectrum analyser when you switch the EQ band in and out.
- A peaking EQ band giving a broad boost in the upper mids, centred around 4kHz. I dialled the Q down a touch from its default setting to make this a gentle, wide boost. With a bit of experimentation, I found that around 1dB gave a nice lift without starting to sound artificial or hyped.
- Another peaking band giving a small boost at the high end, this time with just half a dB of gain and an even wider Q. This helped to give the ride cymbals a little extra sparkle but without becoming harsh. The two wide boosts joined together to make a long gentle curve rising from around 1kHz.
In the end I used bands 1, 3 and 5 (highlighted in blue). You can see where I experimented with other bands that in the end I chose to bypass. (An advantage of having multiple bands of EQ in one plugin is that you can set up a few different cuts and boosts, then switch them in and out to compare.)
I also used CTRL-click (right click on a PC) to open the context menu and enable oversampling, which can help make EQ8 sound slightly smoother in the high end.
Checking the overall output levels with the EQ in and then bypassed, the difference was negligible, so there was no need to adjust the master gain in EQ8 to level match.
Let’s listen to how this sounds. Here’s the original premaster again:
And now the EQ’d version:
The differences are very subtle. If you listen from around 30 seconds, you should be able to hear how the ride cymbals sparkle a little more in the EQ’d version.
Now would be a good time to save the file, if you’ve not done so already.
At this point, mastering engineers would often add other processes such as compression, multiband compression, possibly some saturation, but to keep things as simple as possible we’ll proceed straight to the heavy lifting: limiting and loudness.
A fast peak limiter will enable us to increase the loudness of our audio, i.e. the average level, in a way that still retains the overall tonal character. It does this by detecting peaks that cross a certain threshold, drastically reducing their level, and then rapidly releasing the limiting action. By squashing the transient peaks down, the overall level can be raised up.
Limiting also has the advantage of keeping the levels strictly below the threshold set, so we can get the peaks close to zero dB (i.e. the maximum possible level in a digital system) without worrying that they will shoot over and cause distortion.
Limiting does have some downsides. By definition, it decreases dynamic range, i.e. the difference between louder and quieter parts of a track. Dynamics are a large part of what creates variation and interest. Used excessively, limiting will reduce dynamics to the point where a track starts to sound crushed, flat or tiring to listen to. There’s a good article about this here from Bob Weston’s Chicago Mastering Service.
The key point is that there is always a trade off between loudness and dynamic range. More of one will mean you have less of the other. So for limiting our self-master, the aim is to get a good balance between loudness and dynamic range.
The stock limiter in Ableton Live is a bit basic for mastering, so I suggest using a third party plugin. I’m going to use FabFilter Pro-L 2, which recently became my mastering limiter of choice because it can sound very transparent, has a number of features that are useful for mastering, and has good metering built in.
Alternatives include Tokyo Dawn Labs Limiter 6 GE, Ozone’s maximiser section, Limitless, PSP Xenon, and Waves L2 and L3. If you’re looking for a no cost option, an earlier version of Limiter 6 is available for free. It’s a little more complicated than FabFilter but works well.
You will also need a loudness meter. Pro-L 2 has one built in, but if you’re using a limiter that doesn’t offer this feature, add a separate loudness meter plugin after the limiter, such as the one that comes with Logic Pro X, or the Youlean loudness meter (a free version of that is available).
The next stage is to pick a loudness target. This will depend on the type of material and the intended use. For example, a podcast will typically be mastered less loud than a pop song. Youlean has a helpful table listing loudness standards for various platforms and streaming services. Loudness is measured in Loudness Units Full Scale or LUFS. As a rough guide:
- A common loudness standard for streaming music is -14 LUFS with a true peak of -1dB. This is used by Soundcloud, YouTube and Spotify, and is not far off the equivalent standards for Apple Music, Deezer and so on. Many of these services now have level control mechanisms that will adjust your music to their preferred loudness, so it makes sense to aim for -14 LUFS if streaming is your primary aim for your self-master.
- For dance music, if you want to include your track in DJ mixes or send it out to DJs, it’s typical for loudness in digital tracks to be a little higher, in the range of -12 to -10 LUFS. To my ears, -10 LUFS can sometimes start to sound crushed, so I would suggest using that as a maximum, unless you deliberately want to make your self-master hammer at the expense of dynamic range.
- If you’re mastering spoken word such as a podcast, a good loudness level is somewhere between -20 and -16 LUFS with a maximum peak of -1dB (this is the AES streaming standard).
Let’s go back to the reference track I’ve chosen and run that through an instance of Pro-L 2 with no limiting engaged to see what the loudness is, for comparison. You could try this using a loudness meter on a few tracks in the style you’re aiming for and make a note of the numbers, to get a sense of what is typical. As you can see in the screenshot below from the bottom right of Pro-L 2, I’ve set the meter scale to ‘loudness’ and the meter time scale to ‘integrated’, which will give us an overall loudness measure for the whole track. Having played the reference all the way through, it’s coming out as -9.8 LUFS. For the sake of simplicity in setting a target, I would round that down to -10.
So let’s now go back to my self-master clip, and aim for a LUFS of around -10. I suggest starting by setting up the peak limiter as follows:
- Set the metering to loudness, integrated.
- Turn true peak metering on and true peak limiting on. I won’t try to explain true peak limiting here – in simple terms it means the limiter will be very strict in ensuring the audio doesn’t go beyond the maximum level set at the output.
- Turn oversampling on to 4x. This helps the limiter to catch the peaks. High levels of oversampling can sometimes introduce softness into the sound, but the 4x option in FabFilter seems acoustically transparent to me.
- Turn lookahead up to around 4ms. Again this makes it easier for the limiter to catch the peaks.
- Adjust the output level just below the meter to -1dB. This will give us a -1dB true peak level, which is a safe maximum level to aim for.
- Under the output level knob, click the DC button. This filters out any DC offset in the audio, which will eat up headroom. In our case this should have been removed by the high pass filter we set in the EQ stage, but to be on the safe side it’s good to get in to habit of turning this on in the limiter too.
- I suggest turning dithering to 16 bit, in preparation for the final output stage. Dithering is a complex topic, too much to get into here – this setting should help avoid problems.
If you’re using a different limiter, just approximate these settings as best as you can.
With the limiter set up, we can now get to work:
- Set the track playing and raise the input gain of the limiter (in Pro-L 2 this is on the left hand side of the plugin window). Bring it up gradually until the meter starts to read somewhere close to the target LUFS.
- Then adjust the limiting style and other parameters while listening very carefully to how the sound changes. For example, I often like the results of the ‘transparent’ style in Pro-L 2, but on this track, with that style selected, as I raised the gain the kicks started audibly distorting. Switching through the other styles, ‘dynamic’ gave a more transparent result.
- I generally find that the attack, release and channel linking controls work well left at their defaults, and that was the case here.
Here’s how the final settings looked:
I found that at -10LUFS the track was starting to sound sightly crushed with the dynamics squashed, so I backed off the gain a little until I was happier. That gave an integrated LUFS of -10.9, which seemed close enough to the original target.
In my experience, limiting can often reveal mix problems. Or it might reveal that you’ve been over zealous with EQ in the previous stage of the mastering chain. If that’s the case, you might choose to go back to make some adjustments at this point. Problems are always better solved in the mix than in the master.
Once you’re happy with how the master sounds, save the project again then proceed to the final output stage.
5. Output the self-master
I recommend outputting self-masters as WAVE files (commonly referred to as ‘WAV’ after the file name extension). The other option for a high quality file type is AIFFs, but anecdotally, DJs I know have reported problems with AIFFs in some systems they use, so that’s worth bearing in mind. Most pro mastering engineers seem to use WAV. The advantage of AIFFs is that they can contain metadata, whereas WAVs can’t, so that might swing the decision for you. Both these file types are large but high quality because no data compression is involved.
You might also want to output a compressed version at this point e.g. as an MP3 or AAC file, depending on your requirements.
In Ableton Live, select the audio of your track in the arrangement window (this will ensure the whole track is output – no more or less). Then go to the file menu and select Export Audio / Video, and adjust the settings as needed.
For the purposes of this tutorial, I’ve opted to export as a 16 bit 44.1kHz WAVE file. Now that we’ve got the dynamic range where we want it with the limiting, there’s no need for 24 bit. 16 is fine, will be more compatible with different devices than 24 bit, and will reduce the file size without any audible loss of quality. I’ve selected no dithering in the export settings, because I set up FabFilter Pro-L 2 to perform this function. I’ve also chosen to output a 320kbps MP3 so I have a version that is more suitable for to uploading to this web page.
So – drum roll please – here is the final self-mastered audio clip (MP3 version):
And here is the released pro mastered version for comparison:
I’m happy with this result. I think it shows that with some careful listening, attention to detail and a good limiting plugin, it’s possible to produce a self-master of sufficient quality for the purposes outlined at the start of this tutorial.
Thanks for reading this tutorial. If you have any comments, questions or suggestions of your own tips for self-mastering, please feel free to leave them below. If you’re interested in learning more about audio production, I also offer one to one tutorials for reasonable prices. Full details here.