The manual for the Reason (available on the manuals volume of the server) has a chapter for the NN-19. It would be good to read this chapter.

Essential to understanding how to program patches in the NN-19 is to understand the hierachy ranging from audio file to complete patch. Working from the bottom up (or beginning to end):

Samples ? Key Zones ? Key Map ? Synthesis Controls ? Real-Time Controls ? Patch

Samples are the audio files on your storage medium (hard disk). You should copy your audio files into one folder for use in all of you NN-19 patches. Edit the audio files to remove unwanted silence at the beginning or end, and/or to focus on specific sections of the file that you want to use in the sampler. You can also use Peak to set loop points, which will be read by the NN-19. It is much easier to edit loop points in Peak than in the NN-19.

Key Zones are containers for audio files in the sampler. You load samples from disk into key zones, which contain information regarding the key range of MIDI input that will trigger sample playback, the root key that triggers playback at the original speed of the recording, tuning of the sample, and loop playback mode.

A Key Map contains all the key zones for a patch. Our patches are focusing on single key zones for a patch.

Synthesis controls are used to determine how samples are played back via MIDI noteon messages. The NN-19 breaks these into sections.

The oscillator section controls pitch and sample start point. Interpolation should always be on. Keyboard tracking should be on if you want the patch to change pitch (by changing playback speed) based on MIDI note number (key) pressed. The tuning controls of octave, semi, and fine show musical pitch values as you change them. Envelope amount controls how much of an Attack-Decay pitch envelope you want applied to the signal. Turning to the right creates an upward pitch shift at noteon, followed by a downward glissando to the sustaining pitch. The decay time is unchangeable, and is proportional to the distance of the glissando set by the amount knob. Moving the knob to the right (counter-clockwise) creates a downward pitch shift on attack, with an upward glissando to the sustaining pitch.

The amplitude envelope section contains an ADSR envelope for controlling overall amplitude of each note, with a level knob that determines the maximum amplitude of the envelope (scales the envelope output).

The filter section contains controls for selecting filter type (low pass, etc.), cutoff/center frequency, resonance, keyboard tracking of frequency, a filter envelope, a knob for setting the amount of the envelope, and an invert button for the envelope. The filter can be turned off completely with a switch (next to the word filter on the front panel). Refer to my post on filters for a discussion of filter types and parameters. The keyboard tracking control allows for changing the cutoff frequency according the midi note number played. (KBD set to straight up should give you a match with the KBD tracking of the OSC section.) The filter envelope is an ADSR type, and is triggered along with the amp envelope at noteon. The amount knob controls the amount of frequency deviation that the envelope will apply to the filter, and the invert switch is used to create downward movement of the frequency based on the envelope.

The LFO section contains controls for picking the waveform of the LFO, setting its frequency (rate), its amplitude (amount), and the modulation destination. The destination is limited to pitch (osc), filter frequency, or panning. Please don’t choose the panning option. You can also select sync, which changes the rate control from frequency in Hz to rhythmic values based on your sequence tempo.

The remaining sections of the NN-19 fall into the Real-Time control category. The first is Velocity, which determines how your patch responds to input key velocity. All the controls have positive and negative directions, meaning that a control can increase in proportion to key velocity, or it may change in inverse proportion to the key velocity. The most usual mapping for key velocity is amplitude, with mapping in the proportional (positive) direction. Another common control is to map velocity to filter control.

The Mod Wheel section maps incoming mod wheel continuous controller data to synthesis parameters in a way similar to the velocity section. Common mappings include using the wheel to change the LFO amount, and changing the filter frequency. Mod wheel to filter resonance can produce musically interesting results as well.

The final real-time control section is labeled Controller, and responds to incoming controller data that our lab MIDI controllers do not output by default.

The final section is unlabeled, but controls note triggering and polyphony. You can be generous with polyphony settings (how many simultaneous notes that can be sounded at one time). The computer CPU can play 30 voices without really sweating. Portamento controls the time it takes to change pitch when a new note is pressed. This control only has real use in monophonic modes (1-note polyphony). The mode section should be kept at the default Key setting. The legato/retrig setting is also a monophonic control, and determines if a new amplitude envelope will be triggered when you press a second key while holding down the first key. Finally, you have a control for setting the semitone (half-step) range of the pitch bend wheel.

In general terms, an envelope is a function that changes over time. Typically, an envelope is a multi-segment function for controlling a synthesis parameter. Amplitude envelopes are most common, but any parameter of synthesis (pitch, filter frequency, lfo rate, etc.) can be controlled by an envelope.

The most typical type of envelope used is the ADSR: Attack, Decay, Sustain, Release.

• Attack time: the time it takes from zero value to maximum value.
• Decay time: the initial decay from the maximum attack level to the sustain level.
• Sustain level: the sustain level occurs after the attack and initial decay.
• Release time: the time it takes to go from the sustain level to zero.

I left out some tips that you should follow to keep track of all your files for the creative project with DP and Reason.

Store your Reason patches and racks in a folder inside of your DP project folder. I created a ReasonFiles folder. It would be a good idea to make subfolders inside of the ReasonFiles folder, so that you can store patches for each iinstrument (and your racks – Reason song files) in separate folders. You should also make a subfolder for your audio files that you use for samples, since these are not saved as part off the patch files.

So the folder hiearchy is something like this:

• project folder (Kothman2 project) that contains the DP data file, DP audio files, analysis files, and a Reason folder.
• the reason folder should have subfolders for each instrument’s patches, and a subfolder for your rack files. It should also have an sample subfolder.
• The sample folder doesn’t have to be subdivided by instrument.

You want to file things this way for several reasons. Since Reason doesn’t store samples, only where to find them, you want to put them somewhere that will get backed up with your project. Use the finder to move them into the subfolder before you load them into a sampler. Another reason to put things into subfolders is that it lets you use the arrow keys by the browse icon to switch between patches for an instrument, and switch between samples in a key zone. Once you’ve loaded a patch from a folder, the instrument looks in that folder first for other patches when you use the arrow keys. It makes for easy patch browsing without having to navigate a browser window.

I’ll talk about this more on Tuesday.

We’ve had an old problem come up – namely, not being able to hear audio input from Reason despite having input monitor enabled.

The problem: you have made the proper connections between Reason and DP, so that you see audio output at the hardward interface in Reason, the proper Reason input is set to a DP stereo audio track, and the track is input monitor enabled, but you can not hear your audio through DP.

The test: record enable the track, along with the midi track that controls that synth in Reason. Play something on the MIDI keyboard. Did MIDI and audio data record? (You’ll know from the graphic waveform display if it did. Double click it to see it larger.)

If the answer is yes, the solution has to do with an audio input monitoring mode selection in DP.

In DP, choose Setup | Configure Audio System… | Input Monitoring Mode…

In the resulting dialog box, make sure that “Monitor record-enabled tracks through effects” is selected.

Rewire and instrument plugins are considered “effects” in DP. You can record their output, but not monitor them in “Direct hardware playthrough” mode. (Direct mode is useful for creating cue mixes in recording sessions without any computer latency.)

This post and subsequent ones with the Basic Synthesis Controls… title break down the lectures and reading (pp. 193 – 198) into easier to digest chunks.

Gain

Gain refers to the amount of amplification applied to a signal. Typically it relates to loudness. Internal to a system it can affect other parameters. Gain can refer to either boosting (+) or cutting (-) the amplitude. Cutting the amplitude of a signal is often called attenuation.

Filters

A general definition for a filter is anything that changes the gain of frequencies in a sound. This definition is so general it is useless, since by this definition all audio producing, recording, transmission, and listening devices (including rooms, ears, etc.) are filters.

Typically, we use the term filter to apply to devices that intentionally control the amount of gain in a prescribed frequency range of a sound. Filters are defined by the frequencies they let through (pass band) or frequencies they reject (stop band). The basic filter types for synthesis are low-pass, high-pass, band-pass, or band-reject (notch).

Low-pass

• allows frequencies below a cutoff frequency to pass through.
• cutoff frequency is defined as being at the -3 dB point on the downward slope of the filter curve.
• the slope is defined by its steepness. A typical filter slope is 6 dB per octave (note difference from text, which goes for 3 dB). Higher slopes produce more targeted filtering, but introduce other potential problems.

High-pass

• allows frequencies above a cutoff frequency to pass through.
• other parameters (cutoff and slope) are same as low-pass filter

Band-pass

• allows frequencies around a center frequency to pass through (or between a low cutoff and high cutoff).
• literally a combination of low-pass and high-pass filters.
• generally address center frequency, rather than cutoff frequencies.
• slopes are added, so a 6 dB slope below the center frequency and a 6 dB slope above the center frequency yield a 12 dB slope for the filter.

Band-reject (notch)

• rejects frequencies around a center frequency (or between a low cutoff and high cutoff), not allowing these frequencies to pass through.
• other parameters same as band-pass

Filter Resonance

A resonance control on a filter controls the amount of signal feedback to the filter. Feedback takes the output of a system, scales the amplitude (down), and mixes it with current input to a system. Feedback will boost frequencies around the cutoff or center frequencies.