Enhancing your internal notebook speakers without using an Equalizer

The most obvious problem of notebook speakers is the tinny, pressure-free sound not reproducing any kind of low-end. This is because of the size of the membranes and the inadequate housing. Fortunately software can work around this issue at least a bit.

Often the highs are quite bitchy, not representing the "air" but torturing the listener with searing sibilants. In order to fix that we need to equal this frequency range out.

Normally the relation between the different frequency ranges low, low-mid, high-mid and height is quite out-of-bounds. This needs correction, although I promised a tutorial without EQ.

Because of the thin and neutered sound users tend to raise the volume or even add an Equalizer to raise missing frequency ranges. This normally leads to digital distortions because every signal above 0 dB at the sound card DAC will go from occasional crackling up to a totally distorted sound.

With movies, but music as well, those tiny transducers tend to wiping away low-level sounds but going ballistic on loud and bassy sounds. For this reason it's recommended to reduce the overall dynamic range so you don't need to press your ear onto the palm rest when the lady in the movie starts whispering.

The stereo base is quite narrow because of the positioning of the speakers inside the laptop which gives an impression of almost mono signals.

We will start with the last signal processor being used. As I explained earlier, as soon as the signal hits the sound card DAC it has to be below 0 dB, otherwise it will result in a distorted, messy sound.

Delete already present processors in the list, then add the Limiter. You will not hear any difference, which is what we expect. It will hit in as soon as we start messing up the signal with all the other processors. Set PreAmp to 0 dB, Lookahead to 4 ms, Attack to 2 ms and Release to 8 ms.

A Limiter monitors the signal in (nearly) realtime and detects peaks over a certain Threshold, in our case 0 dB. Set also Stereo Link to 100%, Oversampling to Half x4, disable External Sidechain and Auto Leveling and leave remaining options to their defaults.

As soon as a peak is detected, it starts lowering the volume constantly (within the Lookahead) until the peak will be below Threshold. Afterwards it raises the volume again (within release time frame) until it is back to normal. This way we avoid signals above Threshold hitting the DAC without being noticed too much. If one tries to overdo limiting, it will definitely result in distortion and other side effects, but in our case it's just a fire-and-forget troubleshooter so we can concentrate on the results without trying to mess with levelling over and over again.

Don't overdo things! This might render your overall sound experience worse than before.

Let's look at the real processing now.

First of all we should remove frequencies that cannot be heard because even if the speakers are unable to reproduce them a lot of energy will be wasted in the process. This will lead to problems like not being able to use the full dynamic range and the introduction of digital and physical distortions.

Add the Filter to the list and place it on top, above the Limiter. Set the type to 36 dB High Pass (this will cut off everything below a frequency and will let everything above it pass) and the frequency to 50 Hz (which is definitely below your laptop's speakers abilities). Continuously raise the frequency by 10 Hz until you start hearing a change in the lower end of the audible signal, which should be around 130-160 Hz. Subtract 10% (or 15 Hz) and set it as the Filters frequency. On my machine it is 145 Hz, since the signal started being affected at 160 Hz.

What happened is that we removed everything the speakers can't reproduce at all, leaving a little bit of signal just below this frequency, which is quite important for the next step.

Let's grow some testicles to the signal.

Add the Bass Enhancer and place it at the second position, between the Filter and the Limiter. Set the Floor to 10 Hz and forget about it (we already removed everything below the Filter plugin cutoff frequency). Raise Harmonics to 10 and start with a Scope of 200 Hz. Now start raising the Amount continuously until the bass takes over the whole energy, which is definitely too much. For me it works best at +24 dB. Which is huge but my ThinkPad P1 has the worst speakers I've ever heard in a 2.8k € notebook.

What happens is that the Bass Enhancer adds frequencies to the signal which weren't there before. While an Equalizer only raises stuff which is already in the signal, a Bass Enhancer improves the lower end by adding distortion to a specific frequency range. This kind of distortion adds so called Harmonics to the signal which is a multiple of existing frequencies. On the Blend slider you can choose seamlessly between 2nd and 3rd Harmonics, which means that a frequency of 100 Hz produces a new signal of 200/300/400/600/800/900 Hz... at varying (due to the Harmonics setting) and constantly decreasing levels. This means we're effectively raising the lower range into something better reproducible. It doesn't add or raise the real fundament (so your trousers won't start to flutter), but makes more audible signals that have been swallowed by the hardware before.

Later on, in order to refine the result, you should play with the Amount parameter to define the level of bass and maybe with the Scope to define the upper end of our additional Harmonics.

This is the most demanding step in the chain. Even professionals have great respect for this signal processor since it's absolutely able to destroy your signals big time. But let's be honest, we're consumers and no one will hear the difference between 100 ms and 200 ms Attack time - even on notebook and tablet speakers. This means we can work with some reliable, well established defaults for the timing and frequency splits.

The Multiband Compressor will solve several things for us. It will cut outstanding and annoying frequencies, lower the overall dynamic range and figure our overall frequency response. The latter is what most people try to achieve with an Equalizer. Thing is Equalizers aren't dynamic which will add to the problems instead of solving anything.

Add the Multiband Compressor and place it at the third position, after the Bass Enhancer and before the Limiter.

Since the Bass Enhancer really raised the energy and level of the signal, lower the Multiband Compressor input gain to -6 dB as a starting point. Enable bands 2, 3 and 4. Set Modern Operating Mode and disable the Sidechain Boost. Then let's set the split frequencies for the different bands:

Afterwards let's set some defaults:

Compression Mode for all enabled bands should be Downward. The remaining parameters can be left to their defaults. These values should be a good starting point.

A Compressor lowers signal levels which appear above a certain Threshold. It adds a simple multiplication to the signal defined by the Ratio. If the Threshold is set to -24 dB and Ratio to 2, a signal of -12 dB would in fact become -18 dB. In this example one can then raise the Makeup parameter to 6 dB, since this is what went "lost". So first of all we're rendering loud signals quieter to then raise the overall volume, which leads to less dynamic range with quiet parts becoming more audible.

The timing settings affect the "speed" of the compression. Attack means that a signal needs to go over Threshold for this amount of time until the Compressor reaches the full (negative) amplification. Release defines the amount of time the signal has to stay below Threshold until the amplification is back to 1 again.

A Multiband Compressor splits the signal in various frequency ranges to work on them exclusively with dedicated compression stages. This means that a very loud bass drum would not affect a crash cymbal, since they are clearly located in different bands.

Start playing around with these parameters:

After figuring our overall sound let's look for some icing on the cake.

Add the Stereo Tools and place it at the fourth position, after Multiband Compressor and before Limiter. This tool has a lot of functionality but we only need just a single parameter. Switch to the "Output" tab and set the parameter Stereo Base to 0.25. That's it.

Stereo Base acts on the two channels. It adds inverted signals crossover-wise. This way one can bring a full stereo signal down to mono seamlessly (by going negative values, -1 is fully mono) but also raise the stereo base by choosing positive numbers. This setting has lots of side effects, so avoid overdoing it! One of the bigger problems is that it reduces the bass range due to different effects, which is something we definitely don't want here.

Play with Stereo Base to spread the appearance of your sound to a nice, broad stage without losing the center (vocals, snare drums, conversations in movies) or substance in the lower end. Most likely values between 0 and 0.5 will give some good results.