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Ryan_Vickers

[Guide] Take full control of your system audio

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Posted · Original PosterOP

Intro

This guide will cover everything you should need to take full control over your system audio, including but not limited to setting an EQ, compressor, VST plugin, or other effects on either an input or output.  I will be using Equalizer APO for this task.  https://sourceforge.net/projects/equalizerapo/  Sadly it is only available for Windows.

 

If you've ever been watching a video and wanted to complain about the audio, this guide is for you.  Hopefully by the end of this you will be able to setup whatever is needed to make it not only listenable but truly pleasant.

 

Note that I will not be guiding you to a single complete setup but rather giving you the knowledge necessary to make your own configs as needed.

 

Installation

Once it is installed and ready to be configured, you'll see this screen asking which audio devices you want to be able to use Equalizer APO on.  Select the one(s) you want.

Obviously my list of devices may vary from yours.  Mind the separate playback and capture tabs if you intend to use it for input devices as well.

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You will have to reboot after this for the program to work.

 

Configuration

At any point from here on out, you can control your settings using the Configuration Editor.  Just search "config" in the start menu:

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You will be greeted by the default list of controls:

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Start by removing everything listed using the [ - ] button if shown, or by expanding the additional options for that effect with the drop down arrows and clicking "remove":

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To add a new effect, use the [ + ] button and choose what you would like.  In this case we will add a 31 band EQ:

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You can move the dots around visually, or control the exact frequency and amplitude in the table to the right.  You can toggle any effect on and off using the power button on the left side:

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If you know exactly what you want, this is useful, but personally I prefer to just stack parametric filters.  Let's add one now:

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Here I have inserted a "high shelf" filter.  Note the EQ area.  The red indicates that we are clipping.  If you were using an analog EQ, (for all intents and purposes) this wouldn't happen - it would simply amplify the treble more.  But here we are working within the digital range.  Just as every subpixel in an (8 bit) image can only be some integer between 0 and 255, a "sample", or sound wave's amplitude, can only be between zero, and the maximum value.  Just as you can't have a pixel be brighter than white, you can't have sound be louder than the maximum representable volume.  Therefore, one of three things must happen if you want to increase the treble (or any frequency):

  1. Clipping.  This simply caps or clips off what would have been the amplified sound wave, and creates a very annoying "ear rape" effect (distortion).
  2. "Pulsing".  This is what I call it.  Basically, if the treble in the signal you are playing is quiet enough that you can stay within range despite the amplification, it will play exactly what you want, but if the treble gets loud enough that it would have to clip, it will instead drop the volume of everything to "make room".  This creates an annoying volume pulsing effect, hence my name for it.
  3. Lower everything else.  The fact is you can't boost above maximum volume, so the way you achieve this is to turn down everything else in software, and then turn up your actual volume to compensate, be that the Windows volume or the volume on your amp, etc.

By default, Equalizer APO used to use method 1 but it now uses method 2.  The built in Windows EQ, if present, has always used method 2 as well in my experience.  Since both of those options are terrible, thankfully it's easy to manually direct it to go with option 3.

 

Simply add a preamp and turn down the gain by as much as the largest boost you want to achieve.  You will know you have the right value with the "Peak gain" is no longer red, or at least reads 0 dB.

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Equalizer APO can do more than just EQ however.  Consider the following scenarios:

 

Audio is lower in one ear than the other

This can be fixed by adding a preamp to just one channel:

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Here you can see we've added a channel selector (Control > Channel) so that the effects below are only applied to that channel (in this case, the left).  Notice the EQ panel showing that the left channel is amplified, but the right channel is unaffected.

 

Alternatively, you can simply flatten everything to mono.  This is particularly useful if one channel is not just quieter but entirely missing.

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Here we have added a channel copier (Basic filters > Copy).  As you can see from the arrows, audio from the left input is sent to both the left and right outputs, as is the audio from the right input.  But what if this isn't enough?  Sometimes there is a situation where one channel sounds fine and the other sounds terrible.  Flattening to mono in this case doesn't really help, but we can take only the audio from the left, for example, and send it to the left and right outputs to get around this.

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Volumes vary wildly from one moment to another

In this case you will want to run a compressor.  A compressor takes any sound louder than a given threshold and compresses it by the given ratio.  Finally, make-up gain is used to bring everything to nominal levels.  In Equalizer APO, the best way to run a compressor is to run a VST plugin (Plugins > VST Plugin). Note that this tool can of course be used to run any VST plugin, not just compressors.  You must ensure the architecture of the plugin matches your system however (32 vs 64 bit).

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Use the Options dropdown to "embed" the plugin so it is visible as shown here

Here we can see I've added the "SAFE Compressor", from here: http://www.semanticaudio.co.uk/projects/download/

The controls I haven't described above include:

  • "knee", which rounds the corner between the compressed and uncompressed ranges,
  • Attack (A), which is how quickly the compressor will respond to changes (cut the volume), and
  • Release (R), which is how quickly the compressor will return volume to normal after a loud spike.

If you set attack and release too low, you will get annoying pulsing of the volume.  If you set them too slow, you may notice a distracting quiet period following every (what would have been) loud event.  Experiment with this for your own content.  As you can see, generally I just set them really low and it's fine.  The trick is to not push the compressor too hard.  This one is set to 20 dB but I have a variety that I will use, and I choose the one that is strong enough to get the volumes I want without creating the aforementioned pulsing.

 

Audio is too harsh or "in my head"

If you're listening to content recorded way too close to the microphone, you can use an impulse response file to add reverb and "push them back" (Advanced filters > Convolution).  This can be used for other purposes as well.

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In all likelihood this will be too strong of an effect though, so you can mix in some "dry" signal with the "wet" by copying to virtual channels:

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Only some of my devices need help

You can toggle these filters on or off at any time using the power button on them, or you can setup Equalizer APO to only apply effects to certain devices using a Device filter (Control > Device)

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Everything that follows will only run if you are using the device chosen.

 

My config file is too long!

You can save a list of effects to any text file and then include it all as one group (see the "example.txt" included by default shown near the beginning of this guide).  This is convenient for toggling many effects at once.  If you make changes to an included file, remember to save it, and toggle it off and on again in config.txt to make sure the changes take effect.

 

Closing remarks

There is so much more that this program can do, but this should give you the tools you need to tackle the basics.  No more will you need to complain about bad audio on videos you watch!  (Well actually you should still complain so the creator knows to do a better job next time, but at least you don't need to suffer from it)

 

As always, I hope this was useful and I welcome any additions or corrections :)

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Posted · Original PosterOP
1 minute ago, kodi41 said:

Ryan will this work at the USB level?

It should work on any input or output device in theory, though I can't personally verify all of them of course :P

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I feel like this will severely decrease the audio quality as so many effects are applied to the source digital signal.

 

Btw the "pulsing" can be a part of music and is put into a mastered track on purpose.

It's called a sidechain or a "punch through" kick, which drops the volume of other samples for an emphasis on the kick (or other sample).

Look at the volume faders^


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Posted · Original PosterOP
2 hours ago, Enderman said:

I feel like this will severely decrease the audio quality as so many effects are applied to the source digital signal.

It only applies what you choose to enable.  Plus, when it comes to certain YouTube videos or other content where the "quality" (subjective experience) is already crap, the ship has already sailed on it being hi-fi.  These techniques can help you salvage what's left for a listenable experience when it otherwise would have been unbearable.

 

The intention is not to listen to music through this (save perhaps for the EQ), but to leave it off unless you're listening to a particularly problematic source (WAN show, etc.) where there are random volume spikes, or audio only on one side, etc.  You can, using this program, fix what the uploader did not and take the listening experience from garbage to good.  It's not going to be great because the source is bad, but it's still a huge improvement.

 

And in the case of the EQ, ideally everyone would have an audio system that sounds good on its own, but not everyone has that luxury and this can help them get more out of what they have.  Any minor objective loss in quality would be massively dwarfed but the subjective improvement in experience you can get by EQing poor equipment closer to an ideal response.

Quote

 

Btw the "pulsing" can be a part of music and is put into a mastered track on purpose.

It's called a sidechain or a "punch through" kick, which drops the volume of other samples for an emphasis on the kick (or other sample).

Look at the volume faders^

I've noticed that.  Not sure why anyone would want to add that intentionally, but regardless my techniques will help people avoid adding it unintentionally which is obviously good.

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Howdy again Ryan, well I tried it, the interface takes some experience to use and there is a learning curve for sure. I think this app is more intended towards editing rather then out right audio quality, but your mileage may vary. It did, however, find every single audio path available in the software I have available and can be used via USB. After I installed it I could notice a difference in the sound.  But hey I love to try new stuff.

 

Keep in mind room acoustics and the speaker that is used at that time. One should strive to match the room and the system as a whole.

 

 

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Posted · Original PosterOP
On 10/8/2018 at 9:36 PM, kodi41 said:

Howdy again Ryan, well I tried it, the interface takes some experience to use and there is a learning curve for sure. I think this app is more intended towards editing rather then out right audio quality, but your mileage may vary. It did, however, find every single audio path available in the software I have available and can be used via USB. After I installed it I could notice a difference in the sound.  But hey I love to try new stuff.

 

Keep in mind room acoustics and the speaker that is used at that time. One should strive to match the room and the system as a whole.

Make sure you weren't using it with the default config as this applies a variety of seemingly random effects.  With everything turned off, there should be no difference.  I'd be curious to know if you can hear one though.

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Rep for Harambe

 

 

 

 

 

(I left a real comment in your status :D)


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18 minutes ago, aisle9 said:

Rep for Harambe

you actually did it xD 


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