If you have not noticed the above issues, make sure you are listening in an accurate listening environment (such as with nearfield monitors) and switch to moderate volume (by increasing the monitor volume in your audio interface.)
Using LinEQ to solve equalization issues
Before you alter the waveform; you should increase the bit depth and sample rate from 16-bit/44.1KHz to 32-bit float/96KHz (or 24-bit/96KHz if your audio mastering software does not support 32-bit float). This process is called up sampling. The primary reason is that you are altering digital audio data so make sure you upsample them to have more space for adjustments. I use Voxengo R8brain for this job.
Just set it to change sample rate from 44.1 KHz to 96 KHz and the bit depth from 16 to 32-bit float. Also set the quality to “very high”.
Now open the completed 32-bit float/96KHz audio in your audio mastering software with LinEQ plug-in. I apply the following EQ settings which are quite an extreme setting:
The 12.5dB boost is meant to solve the lack of low end, I attempt a lower gain settings but it’s not just enough. I also cut around 250Hz to remove some muddy sound in the mix and cut a little 5000Hz to reduce sibilance of the mix. I based the EQ settings gain and Q based on what I hear, and make sure you are working with accurate monitoring environment.
The method I use is “Low Ripple” because the EQ settings are a bit of extreme. Low ripple can reduce the side lobe formation of EQ bells adjustment according to the LinEQ manual from Waves. This will give a cleaner and natural sounding EQ result.
Using L2 to maximize the volume
Before you will apply L2 limiter, it is best to convert the sample rate back to 44.1 KHz so that it will be easy for you to apply limiting and dithering all at once. Process the LinEQ output to Voxengo R8brain by converting the sample rate from 96 KHz to 44.1 KHz. Do not change the bit depth; use the same bit depth such as 32-bit float before and after sample rate conversion.
Load up 44.1 KHz, 32-bit float audio (after sample rate conversion) to your mastering software. Then compute the average volume of the track before limiting. In Adobe Audition, I simply go to Analyze – Statistics. Get the Average RMS power; this is the average volume; supposing it is -24.3dB. If I am targeting a -13.5dB after limiter, the threshold on L2 will be:
Threshold= Average volume before limiter + 13.5dB
Threshold = -24.3dB + 13.5dB = -10.8dB
Finally the L2 limiter settings below (I use the high resolution CD master presets with threshold set as computed above):
This is the final master in 16-bit/44.1KHz. Take note the big difference in low end punch, mid and high frequencies as compared before:
Mixing credits by: Coach Janeen,
@SCA Varsity Cheerleaders
Content last updated on August 5, 2012