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Why you should record in higher resolution

I always ask my clients to record or export their tracks in better than 44.1 kHz 16 bit resolution, preferably in 48 kHz 24 bits. A very common reaction to my suggestion is "why the hell should I bother with this if the final master will be in CD audio format anyway?". Here come the reasons.

The quality of a digital audio recording is basically determined by the sampling rate and bit depth used during recording. If you want to reproduce an analogue sound source digitally whose maximum frequency ranges up to 20 kHz (the upper limit or human hearing), you'll have to use a sampling rate of 40 kHz at least (see the Nyquist–Shannon sampling theorem for more info). Since no A/D converters are perfect 44.1 kHz was chosen as the standard in many applications, including CD audio. This means that your A/D converter will capture 44,100 discrete samples from the continuous analogue signal per seconds. Although 44.1 kHz is theoretically enough for perfect reconstruction of an analogue signal I still prefer working at 48 kHz for higher accuracy and preserving more high frequency content.

Now let's see bit depth. Bit depth can be referred to as the resolution of a discrete sample. The CD audio format can contain up to 16 bits of information per sample. This determines the maximum dynamic range and signal-to-noise ratio. In 16 bit audio this is theoretically 96 dB, in 24 bit audio it's 144 dB. So when for example compression or equalization is applied during the mixing and mastering session you further limit the dynamic range and lose bits, i.e. you degrade audio quality.

A good example to demonstrate the negative side effects of this:

I took this high resolution image (click for the larger version):



First I applied some color and tone corrections on it (just like equalization in audio) then I applied a "cut out" filter which reduced the details with a nice character (similar to compression). Here's the result (click for the larger version):



What would happen if the image were resized before applying the corrections and filters on it? Take a look at this picture (click for the larger version):



Comparing it to the previous one you'll find it blurred and less detailed. And that's the point.

If you record your tracks in higher resolution much more detail and transparency can be preserved during mixing and mastering. So when your song is converted to the lower resolution target format in the very last step you'll lose much less from the sound quality than if it were recorded in CD audio format right in the beginning.

Saving a mix containing weak mp3 samples

I worked with a DJ who remixed an early 80's song. Unfortunately there were two transitions from the remix to the original song which was a low quality, (practically) mono mp3 file. So when the playback got to the original sections the previously wide and dynamic sound collapsed, it sounded thin and grainy.

First I tried to revitalize dynamics and match the frequency curve of the different parts. Then I split the original parts to 3 frequency bands and applied a short 10-12 ms left-right channel delay on the highs. And so it sounded awful... Although this technique often works fine with individual tracks this time it did not. The original song parts did not have enough clean air to work with and even worse, the instruments and the vocals just did not come apart.

I thought it once again and found a very simple yet wonderfully working solution. I routed the original parts to two different busses. On the first buss I made the required dynamics and EQ adjustments so the samples sounded bigger and cleaner. On the second buss I set up a tempo synced stereo delay 100% wet, and processed the echos to sound smooth and clean with a weak low end. I turned the volume fader all the way down and then started to slide it upwards very slowly. At a very low setting the mix started to sound quite fine. The original samples had the same punch as the newly produced tracks and a nice stereo width without hearing any echos distinctively. Job done. :-)

Creating a wide natural sounding space

Last week I attended an open-air concert held on a square in the downtown. There were sections when only the vocalist sang or the strings played and I was amazed by the deep and wide space around their main sound. I wondered how I could recreate the effect in the studio.

First I observed the surrounding physical environment. The whole concert area was about 2500-3000 m2. On the left and right sides there were long and high houses with flat walls. In front of me there was the stage and behind me a huge medieval church with four big towers.

After the visual observation I tried to identify the echos I heard distinctively and to match the sound character of the reverberation to a digital algorithm in my head. Finally I came up with the following 3 FX bus setup:

1. 100 ms 3D delay slowly moving between 110° and 200°
2. 120 ms 3D delay slowly moving between 70° and 340°
3. Tempo synced stereo delay (1/8 and 1/16 dotted) panned hard left and right feeding a 2.4 sec hall reverb

I also adjusted the high- and low pass filters to taste.

Curious how it sounds? Download the dry and the processed examples.

Advanced bassdrum(s) management

Recently I mixed a minimal techno track. It had a fairly massive low end originally, but whatever I told to the client she insisted on keeping it very hot. Finally I managed to convince her to at least make two different versions: one for public use and another one for her own pleasure with low end overdose.

She really wanted to feel the kick: she used 5 different bassdrum sounds simultaneously! I suppose this was because her speakers might have lacked low end energy and without knowing proper techniques she wanted to compensate it this way. Things went fine until I got to the 3rd bassdrum track. When I added it, a significant portion of the low frequency content disappeared. Of course, it was a phase issue but after adding the 5th bassdrum, inverting the problematic tracks' phase was not really the solution. It either sounded muffled or hollow, boomy or distorted.

What I did was that I categorized each track based on their main character:

• bassdrum1 - sub
• bassdrum2 - snappy
• bassdrum3 - punchy
• bassdrum4 - dim
• bassdrum5 - big and distinctive

I chose bassdrum5 as the main bassdrum sound because it sounded quite ok and the most powerful. I put bassdrum4 underneath with a gentle low shelf roll off. This gave a slightly different character to the overall sound but without emphasizing the low end too heavily. Next I picked the subbass track with some EQ adjustments to give the mix those deep, floating subs. Finally I routed bassdrum2 and bassdrum3 (the snappy and the punchy ones) to a separate bus with a steep low cut filter and moderate compression to tune their dynamics together.

The final result was a cleaner bassdrum sound with a pleasant, big bottom and still enough edge in the mids to fit smaller speakers too.

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