Polar Response Tutorial
Download the Polar Response (ZIP 2.28MB) file, which contains the three mp3 audio files for this tutorial.
Or listen to the files by clicking on the links below.
Polar Response Tutorial Audio
If you'd like to Do-It-Yourself, then...
Pull out several mics from each operating type and plug them in (if you don’t have several types, maybe a friend has a few you can borrow). Try mics with differing polar patterns (or switchable patterns) and listen for yourself.
Phase Tutorial (Sound & Hearing)
Whenever two waveforms (having the same frequency, shape and peak amplitude) are completely in-phase (meaning that they have no relative time difference) are added, the resulting waveform will have the same frequency and shape... but will double in amplitude. If the same two waves are combined completely out-of-phase (having a phase difference of 180º), they will cancel each other out when combined.
- Load the 0° file onto track 1 of the digital audio workstation (DAW) of your choice, making sure to place the file at the beginning of the track, with the signal panned center.
- Load the same 0° file again into track 2.
- Load the 180° file into track 3.
- Play tracks 1 and 2 (by muting track 3) and listen to the results. The result should be a summed signal that is 3 dB louder.
- Play tracks 1 and 3 (by muting track 2) and listen to the results. It should cancel, producing no output.
- Offsetting track 3 (relative to track 1) should produce varying degrees of cancellation.
- Feel free to zoom in on the waveforms, mix them down, and view the results. Cool, huh?
Harmonics Tutorial (Sound & Hearing)
The presence of frequencies that are higher than a note’s fundamental frequency, help us to differentiate between various instrument types or instrument “voicings”. These partials or overtone frequencies (which are whole-number multiples of the fundamental frequency) are called harmonics.
- Load the first-harmonic A440Hz file onto track 1 of the digital audio workstation (DAW) of your choice, making sure to place the file at the beginning of the track, with the signal panned center.
- Load the second (880Hz), third (1320Hz), fourth (1760Hz) and fifth (2200Hz) harmonic files into the next set of consecutive tracks.
- Solo the first-harmonic track, then solo the first- and second-harmonic tracks. Do they sound related in nature?
- Solo the first-harmonic track, then solo the first- and third-harmonic tracks. Do they sound more dissonant?
- Solo the first-, second- and third-harmonic tracks. Do they sound related?
- Solo the first-, third- and fifth-harmonic tracks. Do they sound more dissonant?
Beats Tutorial (Sound & Hearing)
Two tones that differ only slightly in frequency and have approximately the same amplitude will produce an audible effect known as beats, which are equal in frequency to the difference between the two tones.
- Load the 440Hz file onto track 1 of the digital audio workstation (DAW) of your choice, making sure to place the file at the beginning of the track, with the signal panned center.
- Load the 445 and 450 Hz files into the next two consecutive tracks.
- Solo and play the 440 Hz tone.
- Solo both the 440 and 445 Hz tones and listen to their combined results. Can you hear the 5-Hz beat tone? (445 Hz - 440 Hz = 5 Hz)
- Solo both the 445 and 450 Hz tones and listen to their combined results. Can you hear the 5-Hz beat tone? (450 Hz - 445 Hz = 5 Hz)
- Now, solo both the 440 and 450 Hz tones and listen to their combined results. Can you hear the 10 Hz beat tone? (450 Hz - 440 Hz = 10 Hz)
Masking Tutorial (Sound & Hearing)
Masking is the phenomenon by which loud signals can prevent the ear from hearing softer sounds. The greatest masking effect occurs when the frequency of the sound and the frequency of the masking noise are close to each other.
- Load the 1000 Hz file onto track 1 of the digital audio workstation (DAW) of your choice, making sure to place the file at the beginning of the track, with the signal panned center.
- Load the 3800- and 4000 Hz files into the next two consecutive tracks.
- Solo and play the 1000 Hz tone.
- Solo both the 1000 and the 4000 Hz tones and listen to their combined results. Can you hear both of the tones clearly?
- Solo and play the 3800 Hz tone.
- Solo both the 3800 and the 4000 Hz tones and listen to their combined results. Can you hear both of the tones clearly?
Tutorial: Proximity Effect (Microphones: Design and Application)
Another low-frequency phenomenon that occurs in most directional mics is known as proximity effect. This effect causes an increase in bass response whenever a directional mic is brought within 1' of the sound source. This bass boost (which is often most noticeable on vocals) proportionately increases as the distance decreases. This effect can be beneficial on certain sound sources, however, if you want to compensate for this effect (which is somewhat greater for bidirectional mics than for cardioids), a low-frequency roll-off filter switch can be used. This switch is often located on the microphone body. If none exists, an external roll-off or equalizer can be used to reduce the low end.
- Pull out omnidirectional, cardioid and bidirectional mics (or one that can be switched between these patterns).
- Move in on each mic pattern type from distances of 3 feet to 6 inches (being careful of volume levels and problems that can occur from popping).
- Does the bass response increase as the distance is decreased with the cardioid? … the bidirectional? … the omni?
Soundfiles courtesy of ArtistPro/CoursePTR, www.courseptr.com/artistpro (Professional Microphone Techniques, David Miles Huber & Philip Williams)
Video tutorial available (click on the Video tab above).
Modulation & Noise
Modulation noise is a high-frequency component that causes sonic “fuzziness” by introducing sideband frequencies that can distort the signal (Figure 15.1). This noise-based distortion is due to the magnetic and mechanical properties of the analog recording process itself, and actually increases as recorded levels rise. This noise is often higher in level than you might expect, and when combined with asperity noise (sideband frequencies that are also introduced by the analog record/playback process) can definitely play a role in what could be called the “analog sound.”Do It Yourself Tutorial: Analog Tape Modulation and Asperity Noise
- Feed a 0-VU, 1-kHz test tone to a track on a professional analog recorder.
- Listen to the recorder’s source (input) signal through the monitors at a moderate level.
- Switch the recorder to monitor the tone from the track’s playback (tape) head. Does it sound different?
Download and/or listen to the following compression mp3 audio examples for this tutorial by clicking on the links below.
Input_monitor.wav: 1kHz tone, listening to the recorder’s source (input) signal
A80_tape_monitor.wav: 1kHz tone played back through a Studer A80 2tk recorder *
5050_tape_monitor.wav: 1kHz tone played back through an Otari 5050 2tk recorder
* The A80 (which was originally owned by Bruce Swedien) appears courtesy of Puget Sound Studios. (www.pugetsoundstudios.com)
Time code recorded onto an analog audio or video cue track is known as longitudinal time code (LTC). LTC encodes a biphase time code signal onto the analog audio or cue track in the form of a modulated square wave at a bit rate of 2400 bits/second.
- Go to the Tutorial section of www.modrec.com, click on SMPTE Audio Example and play the timecode soundfile. Not my favorite tune, but it’s a useful one!
- The 80-bit timecode word is subdivided into groups of 4 bits (Figure 11.5), whereby each grouping represents a specific coded piece of information. Each 4-bit segment represents a binary-coded decimal (BCD) number that ranges from 0 to 9. When the full frame is scanned, all eight of these 4-bit groupings are read out as a single SMPTE frame number (in hours, minutes, seconds and frames).
A compressor can be thought of as an automatic fader. It is used to proportionately reduce the dynamics of a signal that rise above a user-definable level (known as the threshold) to a lesser volume range.
The use of compression (and most forms of dynamics processing) is often misunderstood, and compression can easily be abused. Generally, the idea behind these processing systems is to reduce the overall dynamic range of a track, music, or sound program or to raise its overall perceived level… without adversely affecting the sound of the track itself. It's a well-known fact that over-compression can actually squeeze the life out of a performance by limiting the dynamics and reducing the transient peaks that can give life to a performance. Here are various examples of how compression can be used and abused.
Listen to the audio examples:
Download the compression_tutorial_files.zip which contains the mp3 audio examples and the jpgs for this tutorial.
A limiter is used to keep signal peaks from exceeding a certain level in order to prevent the overloading of amplifier signals, recorded signals onto tape or disc, broadcast transmission signals, and so on. Most limiters have ratios of 10:1 (above the threshold, for every 10-dB increase at the input there will be a gain of 1 dB at the output) or 20:1, although some have ratios that can range up to 100:1. Since a large increase above the threshold at the input will result in a very small increase at its output, the likelihood of overloading any equipment that follows the limiter will be greatly reduced. Here are various examples of how limiting can be used and slightly abused.Do It Yourself Tutorial: Limiting
Download and/or listen to the following limiter mp3 audio examples for this tutorial by clicking on the links below.
Delay and regeneration of sound over time is an important effects category that can be used to alter or augment a signal. These time-based effects often add a perceived depth to a signal or change the way we perceive the dimensional space of a recorded sound. Although a wide range of timebased effects exist, they are all based on the use of delay (and/or regenerated delay) to achieve such results as time-delay (or regenerated echoes, chorus and flanging), as well as reverb.
- 16ms comb delay
- 30ms double delay
- 120-189ms stereo delay
Download (or click on and listen to) the following file for this tutorial.
By varying program and setting parameters, a digital reverb device can be used to simulate a wide range of acoustic environments, reverb devices, and special effects. The following examples have been included using the Universal Audio UAD-1 card’s DreamVerb plug-in, available at uaudio.com.
- Hall Cathedral
- Room — Large Dark Room
- Room — Small Chamber
- Plate — 140 Dark Plate
- Plate — Large & Bright
- Drum — Live Room
- Guitar — Solo Presence
- SPFX — Party Sound System
- Download the Reverb Types zip file, which contains the mp3 audio file for this tutorial. Or listen to the file by clicking on the link below.