Cutting and Labeling

The cutting process is long, tedious, and frustrating because even after noise reduction the noise floor is much higher than I’d like.

For each note I have to find the onset, and then cut it at the closest zero crossing.

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I also have to find the end of the decay, which including the reverb, is subtle and can be difficult to ascertain. Again once I find the end point I have to cut it at the closest zero crossing.

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Once I have all the notes cut, I listen to them for any clicks and pops, meaning that I missed the zero crossings, and to make sure that I have all of the notes in order, with no doubling, or messed up notes. After that’s all said and done I can start labeling. Despite the samples being in different keys, I am labeling them in the spirit of the organ, where each stop, no matter the octave, is centered on C4. With the 61 Key manual that means that samples range from C1 to C6, in the bastardization of scientific pitch notation I am using to label my files. This system equates the physical location of the key on the manual to the label, as opposed to the pitch.

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I’m not going to make any excuses, there is one reason why I am not going to upload my files tonight, I don’t feel it’s good enough for submission. This was caused by a combination of things, particularly my poor scheduling, in regards to balancing this, other projects, and work, but also because I made some early mistakes in the process (nosie floor, and sample length) which have been giving me trouble for a while now, but in my stubborness, I’ve not admitted it.

The noise floor is making it so that there is a noticeable dropout between notes, when played through headphones. This could possibly be fixed if I were to create a VST from scratch which filled the in-between notes with the room tone of the church. But that’s something that’s over my head.

Also, because I recorded long enough samples to make looping easy, I am having trouble making notes which are not held for at least the recorded duration of the sample use the natural decay recorded in the sample. I may have to edit shorter versions of the note and rely more on looping.

The point is, my files won’t be uploaded until Friday evening. I take full responsibility for my lateness, including any penalties that come with that. There will also be two more blog posts which chronicle my process, including the editing of the samples and the patch creation, which should go up around the same time as my files.

I apologize for this, and regret not being more open with these problems.

Noise Reduction

So, as I learned during my recordings, church acoustics apply just as much to the heating system as it does to choirs and organs, which means that I have to put the actual creation of my sample patches off, and focus on getting as clean a signal as possible without reducing the fidelity of the organ. Which leads me to look into the process of noise reduction.

I’m going to preface this by saying, I don’t fully understand this process, these algorithms, or half of what I’m going to be trying to talk about in this post.

So, the process of noise reduction can be done in many different ways, but the first thing to understand is the two types of noise reduction algorithms: Fixed and Adaptive.

Fixed Noise Reduction

Fixed noise reduction, or spectral noise gating uses Fourier analysis of sample sections of noise to create a spectrum graph, which is used to gate the audio, reducing the level for any sound that isn’t about the threshold. This threshold varies at different frequency bands in order to sufficiently remove the background sound during sections that are above the threshold.

XNoise – Waves Plugin

This works best for samples which have a consistent noise throughout the entire sample, as the filter is static.

This gating is often combined with other processes such as frequency smoothing and time smoothing, which both are baffling to me for the time being. From what I can gather, these processes help to make it so that the effects of the noise gating don’t degrade the quality of the frequencies that are above the threshold at any given time, but again, I’m still not 100% on what they are or what they do, and definitely not how they work. [1]

Adaptive Noise Reduction

Adaptive filtering is a process which models the relationship between the input and output of a filter throughout the duration of it’s use. This means that it adjusts, or adapts, to the changing signal and as such, adaptive noise reduction is typically used for samples with noise that varies over the duration of the recording.

where x(n) is the input signal to a linear filter

            y(n) is the corresponding output signal

            d(n) is an additional input signal to the adaptive filter

            e(n) is the error signal that denotes the difference between d(n) and y(n). [2]

The process is similar to fixed reduction, but instead of using a Fourier analysis of the noise, the filter is created with a Least Mean Square algorithm. This takes a fixed filter like the Fourier example used above and changes the variables of the filter (ie. the threshold in each frequency band) in response to the input signal.


The quality of both of these types of filtering depend on the specific algorithm used. And quality noise reduction software isn’t cheap. Waves noise reduction plugins range between $200 and $600, iZotope’s RX3 runs $1200 for a full version copy, and CEDAR’s noise reduction hardware units (like the DNS1500) run upwards of $5000.

And it makes sense, in a world where audio is often second (or third) fiddle, it’s not always possible to get the best recordings, so noise reduction is a necessity.

Further Reading

Recording Process

The recording process was both easier and more difficult than I thought it would be. In my infinite foresight, I failed to confirm that I had assistance during my session, requiring me to transport, set up, and recorded solo.


I used an X/Y stereo mic setup, as previously described, running it into ProTools through the MOTU 896 at 96k. I used the Earthworks SR30 small condenser microphones, which are cardioid microphones with exceptional response.



I placed my microphones in the pews, where I started with facing them towards the front of the congregation, but due to the (suprisingly) low level of the organ, I re-positioned them facing the pipes. I’ve already spoken about the awful room noise I experienced. I knew it was going to be a problem while I was in the space, but after 45 minutes of trying to compromise a position between accurate sound of the room and lower noise, I resigned myself to the fact that I couldn’t position out the hiss, hum, and rumble of the room.


Because the recording setup was so far from the manuals, before each stop I had to start recording and run upstairs to the manuals, leaving about 40 seconds of pre-roll on each file. Once I got to the manuals, I played each note for approximately 8 beats, or 2 bars, at somewhere around 120bpm.


There were some oddities to the organ which I noticed as I was working, one being that certain stops had leakage, where the stop being open inherently produced a low level tone from air leaking into certain pipes.This, like the noise floor, wasn’t something I was able to fix with my limited time and inexperience with organs. I also noticed that certain notes had slight variations in the attack, which are hard to describe other than to say that they were dissimilar to the other notes. After discussing my session with Gordon, who is the caretaker of the Organ, he told me that the difference in attack was caused by dust, and that the organ was over-due for a cleaning. Again, there wasn’t too much I could do about it at the time.


In conclusion, the problems with the noise floor, the air leakage, and the inconsistencies of the attack, leave me to characterize my sample gathering as somewhat of a disappointment, if not an outright failure. The only saving grace being that the recordings are incredibly true to the space, and the organ, even if that means that they don’t sound as crisp and clean as I had hoped.


The Stereophonic Zoom – Michael Williams

The Stereophonic Zoom is a document which describes a variable dual microphone system for stereo recording. The system described acts as a spacing unit which can alter the distance between microphones as well as the angle in which they face. The interaction between distance and angle creates a variable stereo width.

Williams argues against the implementation of a singular system for stereo recording because “rather than reduce the choice of systems, an effort must made to increase the number of systems available. Each sound recording engineer must have the largest possible selection of systems to choose from, in order to solve the specific problems presented by a particular recording situation and, to express his own personal interpretation, as freely as possible,”.

He describes the accepted standard listening condition pictured below:


Williams continues by describing the importance of the listening environment in determining the characteristics of the stereo width of a recording. He makes recommendations for treatment following the IEC guidelines for a standard listening room.

Williams goes on to describe how sounds are localized, which I describe throughout my comparison of stereo mic techniques, but include timing differences, varying the intensity between two speakers, or a combination of the two methods.

As per my earlier postings, I opted for an X/Y mic setup, meaning that I’m relying primarily on amplitude variation between the two channels for my stereo.

Williams goes on to describe the specs of the “stereophonic zoom” recording device in great detail, indicating distance v. angle, frequency responses, and the relation between direct in reverberation sound through graphs, and describes the test phases and limitations of the device during testing. This section, while interesting has little to do with my way forward, and can be summarized by saying, on paper this system looks good, though like all things audio, it’s subjective and I’d have to hear results before I’m sold.

Sound on Sound: Synth Secrets

In this series of articles, Sound on Sound writer Gordon Reid writes extensively on how analogue synthesizers work. Parts three of the sixty-three part series concern envelopes, what they are, and how they work. Envelopes are important to my process, because while the envelopes of my samples are “baked in”, the principles of attack, decay, sustain, and release will all apply. With this in mind, Reids article, while fascinating, wasn’t particularly useful to me past his very brief discussion of ADSR (Attack, Decay, Sustain, and Release), and his definition of an envelope.

Reid defines an envelope by saying “the graph of the way a parameter changes over time is a visual representation of its envelope,” (Part 7)

The article defines the four parts of ADSR:

  • Attack: The speed at which a sound reaches it’s maximum volume
  • Decay: The speed at which the loudness drops to the sustain level
  • Sustain: The level the loudness maintains until the release
  • Release: The amount of time it takes to decay from the Sustain level to silence

Interestingly enough, Reid describes the ADSR graph of an organ, because of it’s simplicity. He compares it to two other envelopes, though the comparison isn’t particularly useful to me. He says “The organ has a rapid attack and maintains its full volume before dropping to silence when the player releases the key” .

Envelope Graphs

While this envelope describes a synthesized organ, the release on my samples will have to be longer, to facilitate the natural reverb of the church. However, the fact that their is no decay will make looping my samples easier, because the the sustain is longer, and easy to define.

Choosing a Sampler 2: Choosing a Sampler Harder

Despite my uncertainty, I’ve settled on the sampler I will be using to build the patches.

Kontakt, while expensive, is feature packed, and highly regarded as one of the top software samplers available. The amount of material available to guide me through the process of creating my patches will hopefully make this process as painless as possible, and was in large part the reason I chose to move forward with Kontakt.

This means that I will be looking into the possibility of trans-coding the kontakt files for use in other samplers, although my plate is pretty full right now and it’s not my biggest priority.