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Bedroom music production, gaming and random shit

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making beats

Translate Sub onto Smaller Speakers

There’s a few things you can do to make your bass sound on smaller speakers like laptops, tablets and cellphones. First you need fundamental and harmonic content on your bass. The fundamental frequency is the base foundation note that represents the pitch that is played and the harmonics are the following frequencies that support the fundamental. In short, it’s the higher frequency harmonics that allow for the sub to cut through the mix.

One idea is to create higher frequency harmonics. The harmonics should be in unison with the fundamental frequency, but don’t contain it. (The harmonics trick your brain into hearing lower frequencies that aren’t really there.) Add a touch of harmonic saturation, drive a little warmth, a little fuzz to help that sub cut through. The harmonic distortion, adds high-frequency information to reveal presence on systems with poor bass response.

Also try to copy the bass to a parallell channel, bitcrush the higher harmonics and cut all lows and mix with the original bass.

If you’re beefing up your main bass by layering a separate, low-passed sine wave at the octave below, perhaps try square (or triangle) to add some subtle higher frequencies that allow the sub bass to translate better than a pure sine wave.

You can also try to EQ the bass. Try to boost the harmonic multiples of the fundamental frequency to hear some definition from the bass sound. And boosting above 300 Hz will bring out the bass’s unique timbral character. Actually, try around 1 kHz (but add a low-pass filter at around 2-3 kHz).

Use high-pass filtering (to clear the low-end frequencies and make room for the intended bass sound), and you can also side-chain your sub bass to keep it from fighting with the kick drum.

When it comes to kick drums you can add a small click noise to help it translate onto smaller speakers.

P.S. There are also plugins that use psycho-acoustics to calculate precise harmonics that are related to the fundamental tones of bass.

New keyboard stand, four tiers, compact living. Early December 2016.

Headroom for MP3

I’ve written about the importance of headroom when submitting your track to a professional mastering engineer, but you should also pay attention to headroom when you do this on your own and when you encode MP3s.

Okay so when the track is mastered at 0 dB (the maximum level for digital audio files) many converters and encoders are prone to clip. Lossy compression formats utilize psychoacoustic models to remove audio information, and by doing so introduces an approximation error, a noise which can increase peak levels and cause clipping in the audio signal – even if the uncompressed source audio file appears to peak under 0 dB.

In Practice

For example SoundCloud transcodes uploaded audio to 128 Kbps MP3 for streaming. In this scenario, use a true peak limiter to ensure the right margin depending on the source material. A margin of -1.0 or -1.5 dBFS should work for no distortion (sometimes -0.3, -0.5 or -0.7 would work, but it’s safer to have greater margin).

https://api.soundcloud.com/tracks/292135058/stream?client_id=3cQaPshpEeLqMsNFAUw1Q?plead=please-dont-download-this-or-our-lawyers-wont-let-us-host-audio

“Hope my haters keep a special place in their heart for me”

https://api.soundcloud.com/tracks/275055998/stream?client_id=3cQaPshpEeLqMsNFAUw1Q?plead=please-dont-download-this-or-our-lawyers-wont-let-us-host-audio

158 BPM.

Here’s my setup as of today. A bit crowded.

Compression Time Again

Compression is an invaluable tool that can be applied to almost any sound. Therefore, here’s a friendly reminder about compression and the settings of attack and release on a compressor.

Most times compression is used to control dynamics and taming peaks to get a smooth, consistent signal. Other times compression is used to add punch, impact, proximity or for tonal control.

Four Settings

There are four settings on most compressors. The threshold controls the point at which compression begins. The ratio is the setting for how much compression is being applied. (A so called limiter is a compressor with a high ratio, e.g. inf:1, that will stop the signal at the set threshold.) Then there are attack and release settings. Attack sets how long it takes to reach maximum compression once the signal exceeds the threshold. And release sets how long it takes for compression to stop after the signal gets below the threshold. (Some compressors feature an auto release, which automatically adjusts the release time based on the incoming signal.)

Attack

Attack controls how much initial impact gets through.

A fast attack time shaves off the initial transient impact, and can make it sound more consistent and controlled. But when gone too far, the sound will lose vibrance and seem more further away.

A slow attack time is letting a lot of transient formation through. The initial impact will come through and the compressor will start to work after that. This can make it sound punchy, big and aggressive, but not very consistent dynamically.

Release

For release time, again there are two options: fast and slow. In general, fast release can render a more aggressive, gritty sound – the initial sustain is sort of brought up, meaning more perceived loudness. But when the release time is too fast, it can sound exaggerated, distorted and bad, and there can also occur some pumping artifacts.

A slow release time will give more dynamic control, more smoothness, but also sound a bit distant. And if overdone with a slower release, the compressor will not release in time for the next hit to come through, and that can suck the life out of the initial impact and sound flat.

Stack Compressors

An effective way to stack compressors is to put the compressor with the fast attack time first and the compressor with the slow attack time second. The first compressor will smooth out those transients and make the initial hits more consistent, while the second compressor, fed by the dynamically controlled signal, will accentuate the initial hits.

Switching DAW: Reason to Live

I’m switching to Ableton Live from Propellerhead Reason. There are several causes for this.

In short, nowadays I’m using mostly hardware synths and Ableton Live is more paramount and flexible when it comes to integrating hardware.

Reason’s core softsynths are still good, but I’m using them less and less, and I’ve grown tired of certain limitations and the workflow. So going to a different DAW is a good remedy for that. And I can still rewire (connect) Reason to Live.

And working with up-to-date “real” plugins is so much deeper and at the same time a bit exhausting.

Although none of these thing are new, the last few iterations of Reason (version 8 and 9) are clearly focused on bringing in newcomers without trying to keep more seasoned users.

For me, switching DAW is both fun and challenging. Of course there’s all this learning to be done. But it’s also fun. And it’s really not that hard, it could just seem a bit daunting at first, but there’s great help online nowadays with countless forums and tutorials. Right now I’m on some kind of trial period, and a lot of time is spent trying to connect and run hardware with software, but I think the new main DAW will be inspiring and push my music productions further.

Add Life to Your Mix

Sometimes when I read about music production and audio engineering stuff I come across ideas that I personally wouldn’t use in my music, but nevertheless could be interesting – at least in theory – and perhaps someone else dare to try.

Here’s one: record your “as is” mix from your monitor speakers, using a couple of microphones, and then blend the recording with your final mix.

This could add vibrance and “realism”. It could of course also clutter your mix if you overdo it. If needed, try to poke the recording to play with the phase relationship.

Recording your mix like this can add some analog imperfection by revealing a little of the studio’s ambient, and the colors of the mics, preamps and monitors would also print this sound layer. And you need not to record in the studio, you could put the monitors in a (non-acoustic treated) reverbant room, or record with an opened window… You get the drift.

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