Mastering tutorial

We provide several video and text tutorials about audio processing and using our plugins for both beginners and advanced engineers. Everything in this chapter is copyrighted unless specified otherwise. However feel free to link these tutorials. Quoting the tutorials is possible if link is following the quotation.

Mastering is the essential last step in music production. A mastering engineer takes a mix and makes it sound right for a particular purpose. Typically, the mix is equalized, so it has all the necessary frequencies. Dynamics are then adjusted, to minimize the differences between softer and stronger parts of the recording, and finally compression is added and a limiter is employed to make the overall sound louder, a method that is now very popular. We will present one of the many ways to do this.

There are also some quotes from the Bob Katz book "Mastering audio" , which is definitely worth reading if you are serious about making music.

Before you start mastering

  • The best mastering is no mastering at all!
    If you think your mix is perfect as is, don't process it anymore. Each step in the processing chain adds extra noise and distortion, digital or analog, it is always there.
  • Get the best mix quality you can!
    The better the mix, the less processing needed at the mastering stage. However, your mixes should not contain compression or be equalized too much. Once something has been changed, it is hard to undo it.
  • Always master at 96 kHz.
    Alternatively, you can use up-sampling, however the quality will always be worse and your mix quality is already degraded from being sampled at a lower rate. Many low-end studios still record in 44 kHz. If this is the case, you should use your DAW (e.g. Wavelab) to increase the sampling rate of the mix to 96kHz for mastering, and then convert it back to 44 kHz after completion, ideally with some dithering applied.
  • Always export your mixes in 32-bit floating point format.
    Audio degradation when truncated to 24-bits is typically not audible, but further processing can exaggerate the defects. 16-bit audio degradation is often audible on good studio monitors.
  • Be aware that your ears will soon think your sound is great even if it is not ;).
    You should always master in a DAW where you can switch between tracks, so you can listen to your current audio, the original, and your reference tracks you have chosen for comparison. Before you start, prepare these in your DAW project (e.g. in Cubase), and choose at least one professionally mixed song (ideally up to 10!) for guidance.
  • The 'Loudness war' is not a good thing...
    Modern pop-songs have almost no dynamic range compared to recordings made 20 years ago. It all started when someone discovered that songs sound better when they are played louder. Modern digital processors are able to increase the volume of songs so much, that when compared to their older versions, they sound many times louder.
    However the dynamic range is sacrificed. You can hear that most modern songs sound the same from beginning to end because removing the transients, in order to increase the volume to these incredible levels, removes dynamics of the music. Therefore, always increase the volume only as much as necessary, not one single dB more! It may sound better, but only for a few seconds!
  • No peaking!
    You will create a chain of effects, with each one performing some operation on the sound and very often this sound may get amplified. Most plugins have a gain control and a peak meter to ensure that the output isn't clipping. It may be in the red zone, but not above, otherwise the sound may get distorted, when bypassing an effect for example.
    Moreover, if the output of every effect has approximately the same level as the input, then you may bypass any effect in the chain to check the sound without it. It doesn't always work, but it can help a little.

1. Manage the stereo field

One very hard task of a mixing engineer is to prepare a room for the instruments. In a good mix your brain should be able to identify each single sound source and place it somewhere in the space. This can be managed by various panoramas, reverbs and delays.

When mastering you should ensure that this depth of field is preserved. You can start by checking if there is a good amount of stereo field and correct it if necessary. MStereoProcessor is a tool designed for this purpose.

MStereoProcessor is a 4-band stereo field enhancer and exciter. Use the Widening parameter to make each band wider, or, more 'monophonic'. It's usually good to keep the bass more monophonic, for example. You can use the Exciter to improve the clarity of a band but do NOT overuse it! Delay widening can be useful to add additional stereo content, but use it only if really needed. The goal is not to make an artifically stereo-sounding output, but to control the stereo content.

The resulting signal in the stereo field view on the right should form a nice vertical ellipse, not too wide, and not too thin. You can always use the MStereoProcessor to compare the wideness, of your reference tracks!

Finally you should check for mono compatibility by using the buttons above the stereo field view. Remember that even in the 21st century your recordings should be mono compatible! When you compare the stereo and mono recordings, the monophonic one loses the stereo content, but it should still have some depth and there should be no significant frequency loss caused by phase cancellations. In extreme yet typical cases, a track may completely disappear when played in mono. Poor mono compatibility at this stage, means you will have little choice but to obtain a remix as in most cases this cannot be fixed during mastering.

2. Balance the spectral content

Spectral content describes proportions of frequencies in the audio. Mixes rarely have good overall spectral content, but to be fair, that is not the aim of mixing! You can usually fix this using an equalizer, however very often these disproportions are not constant and are changing during the song.

Spectral content affects the overall loudness a lot. Our brain adaptively masks silent frequencies in order to let us listen to what is important - what is loud. Therefore the loudest recording has equal power over most frequencies consistently (note: this is heavily simplified!). There is an extreme case - white noise. It is a signal, that has the same magnitude for all frequencies. I bet it is louder than any of your recordings! Keep in mind, that every dB you increase in loudness is lost somewhere else.

Spectral balancing and other problems are generally fixed using multiband compressors such as MMultiBandDynamics. They will also reduce the dynamic range. Generally, you use equally distributed bands, about 4 or 5 is usually enough. Set the threshold of each band to a similar value, and the ratio to about 2:1 and then tweak all of the bands. Remember, the less you change the better.

We also have a more modern solution - MSpectralDynamics. At first look this tool may seem overcomplicated, but in reality it's much simpler and safer than using multiband dynamics of any kind.

In the spectral view you can see the power of each frequency. In an extreme case this would form a horizontal line. For starters use the orange line to manipulate the Threshold of Processor 1. The lower it is, the more dynamics you lose and the more balance you get. The next important parameter is the Ratio of Processor 1. We'd recommend starting with one of these settings:

  • Lower threshold and a very low ratio (e.g. 1.10:1) - this would apply a mild compression along the spectrum.
  • Lower threshold a little under problematic peaks and higher the ratio (e.g. 2:1) - this lowers louder frequencies and makes the quieter ones more audible.

After that you can experiment with the various parameters or presets. You may be surprised at what was not audible in the mix before processing. But be very careful not to overuse this tool! If you don't need it, don't use it!

3. Compression.

As we've mentioned already, the modern trend is to over-compress recordings in order to make them loud from beginning to end. So, please read the following advice and remember - if you don't need it, don't do it!

If you are new to this, it will take a lot of time before you will be satisfied with the results. A certain amount of experience is necessary.


Your first aim is to make your song sound consistent - ensure each refrain is not much louder than a verse etc. The best way to do this is using envelopes in your DAW to manually manipulate the gain. But if you are as lazy as I am :), you can start by adding MDynamics to the processing chain with following parameters:

  • Attack time: longer, let's say 100ms, we want to keep transients.
  • Release time: longer, let's say 1000ms, we don't want MDynamics to think the chorus ended just because there is a short silence.
  • RMS length: longer, we don't want it to respond to peaks, but rather an overall level.
  • Threshold: play the verse and then the chorus. The vertical line indicates current level. Place the threshold for Processor 1 (the only enabled) just above the level it measured in the verse. The verse will now be below the threshold, and therefore not processed, while the refrain will be above, hence the gain will be reduced.
  • Ratio: the higher the ratio is, the more reduction and loudness you will get, and, the more dynamics you will lose. Set it at 1.5:1 for starters, then play the chorus and the verse to check if they are similar enough.

Note the Maximize option, which can make the result exceed 0dB. Be aware that you can use the Output gain to ensure that the result does not peak.


You may want to remove peaks and make it sound louder. We repeat again - if you don't need it, don't do it! The loudness will be increased in the final part aswell! However, if you want to try, create another instance of MDynamics with following parameters:

  • Attack time: short, let's say 10ms.
  • Release time: short, let's say 100ms.
  • RMS length: short, but probably not minimal (peak).
  • Threshold: overall level should be pretty consistent due to the previous macrodynamics stage. So play the chorus and set the threshold slightly below the current level (vertical line).
  • Maximize: in this case you may want to disable it. Anyway you will have to use Output gain to ensure the result is not peaking but is loud enough.
  • Ratio: as before, the higher the ratio is, the more reduction and loudness you will get, but the more dynamics you will lose. Set it at 1.5:1 to begin with. Then play the chorus to check for distortion.

You could also use a multiband compressor such as MMultiBandDynamics. It would theoretically be more transparent, avoid pumping a little better and balance the spectrum, but it has a million parameters and is easier to overuse. However, if you want the best quality and have the time, then go for it.

4. (Auto)Equalization.

This will finally make the spectrum sound professional. This is the most important part. Even the compression and limiting stages are expendable, but this has to be present.

Some of you are may wonder why we have put the equalization after the dynamics. This is because the compression may change the spectral content, so we feel it's prudent to put the equalizer here although it is your choice. Probably the best solution is to place another equalizer before the dynamic processing, however we want to keep this tutorial simple.

1. Load MAutoEqualizerLinearPhase, start playback and press Analyze target. A curve will be shown but leave the playback enabled until it stops moving. It is a good idea to play different parts of the song, so the analysis is objective, however the loudest part of the song is the most important as usual. After the analysis has finished, press the button again to stop it.

2. The next step is to get a comparison of the track that you want your recording to sound like. You can either use the Analyse source button and follow the same process as in step 1 above now using your comparison song instead as a source, or simply load a predefined comparison using Comparison / Load.

3. Click Equalize. The plugin adjusts the parametric bands for you. It is probable at this stage that the result may sound a little unnatural. This is due to the fact that you are not yet accustomed to it. Also that the comparison may not be ideal and you may have try another one. However, you can start by lowering the Dry/wet ratio until the result sounds good and natural. Usually 30-50% is reasonable. Remember to check the Gain parameter to ensure the result is not peaking.

Until this point you have been working with a song with very different spectral content, and your ears have adjusted to it. You now have to switch between tracks a lot, mostly between your song and the comparison, in order to give your ears time to be able to compare the recordings objectively. If anything is wrong, don't hesitate to use the equalizer manually, as you can always perform the autoequalization again!

5. Limiting.

And here we are. By now you should have a nice sounding master, the last thing we can do is to make it sound louder, hence "nicer" :).

1. Play the loudest part of the song. Load MDynamicsLimiter. Move the Threshold to 0dB if it's not there already. Now there is no limiting, so only clipping or saturation is performed. Watch the peak meter and use the Input gain to lower it if necessary. The input should not be peaking!

2. Decrease the Threshold very slowly to the point where the peak meter touches the 0dB limit. Not a single dB more, unless you want a crunchy master. If you make the threshold too low, the output will get distorted.

The meter above the peak meter is called the gain reduction meter and shows how much of the track dynamics you have lost. It should be tapping -6dB at most. Finally you can use Saturation to carefully increase the loudness even more. Alternatively, you can use MLimiter to perform the saturation, but in that case, be even more careful.

You can get more transparent limiting using a multiband limiter, but be extremely careful!

Multiband limiters such as MMultiBandLimiter or its simplified version MUltraMaximizer can provide a higher level of loudness than single-band limiters just by increasing the Input gain parameter. Because each band is limited separately, by increasing the input gain you also balance the spectrum and get closer to the white noise we talked about above. This tricks the brain into thinking that it sounds better.

Working with a multiband limiter is not too difficult. Just increase the input gain and watch the meters on the right, especially R, the gain reduction meter, and S, the saturation reduction meter. Saturation reduction causes distortions, and gain reduction causes pumping. Finally, when you think some bands are affected more than others, you can use separate thresholds or band input gains for them.

3. Listen closely to the results and if you hear any unpleasant distortions, clicks or pops, increase the threshold (decrease drive) or remove saturation. If none of this helps, decrease the input gain. If there are still artifacts present, bypass the limiter as it is very probable that the distortions are generated by any of the previous steps (probably dynamics) or are even contained within the mix!

4. Lower the ceiling parameter to say -0.2dB. This is basically just output gain, but some media (mp3 encoding) can by its nature create output levels after decoding higher than the original. So despite we have room to 0dB, it is a good idea to keep the output slightly lower.

And finally, the golden rule as usual - listen. Switch between tracks and compare. Don't make your recording louder than the comparison, there is too much to lose. Compare it to the original too, to see if the mastering has helped. And don't forget to listen to the quiet parts of the song as well, since the processing might also have amplified any noise etc.

6. Exporting the result.

When your master sounds good, go away, grab something to eat, watch some TV, and ideally go for a walk to clear your head. When you get back, which should be at least a few hours later, listen again. Still sounds good?

Great! Let's proceed to the export. Use the export/mixdown feature of your host to generate a wave file at the same audio quality, preferably 96 kHz and 32-bit floats. At this stage you can add some fade-ins/outs if necessary.

Now you have the finished recording at the highest quality. Use your DAW to create a file in the format you need. In most cases you will be down-sampling to 44.1 kHz and decreasing resolution to 16-bits. Dithering is recommended here. You probably won't hear the difference, but when your recording is played on a big concert system, someone might!