Chapter 21 concludes by speculating on future developments in musical applications of microprocessors at all levels from novelties to serious musical research. My custom synthesizer work. Contribute to rabidaudio/synthesizer development by creating an account on GitHub. Musical Applications of Microprocessors [Hal Chamberlin] on kungranaleapu.tk * FREE* shipping on qualifying offers. Chamberlin, Hal.
|Language:||English, Dutch, Japanese|
|Genre:||Politics & Laws|
|ePub File Size:||21.78 MB|
|PDF File Size:||10.55 MB|
|Distribution:||Free* [*Sign up for free]|
Get Instant Access to PDF File: #11f9d4 Musical Applications Of Microprocessors By Hal Chamberlin EBOOK EPUB site PDF. [EBOOKS] Musical Applications Of kungranaleapu.tk Book file PDF easily for everyone and every device. You can download and read. Files Photo Music Books Video Games belajar word pdf hal chamberlin musical applications of microprocessors pdf common as air hyde.
The side-chain is the hand that controls the volume. Side-chain circuitry examines the input signal or a separate Key Input and issues a control voltage to adjust the gain of the signal path. In addition, a side-chain loop allows patching in filters, EQ or other processors to this path.
More on this and a description of the type of side-chain controls later. Some dynamics processors make the side-chain control voltage available for connecting to a neighboring unit, or to tie internal channels together.
Slaving or linking dynamics processors causes the units to operate simultaneously when only one unit or channel exceeds the threshold setting. This feature preserves stable stereo imaging and spectral balance. All dynamics processors carry out gain control as a function of side-chain level.
Some use the internal signal as shown in Figure 1 and some use an external or Key Input as shown in Figure 2. The only difference between a compressor, limiter, AGC, de-esser, ducker, or gate, is the type of side-chain detector, the gain computer attributes and the type of gain control element used.
Figure 2. Dynamics processor with side-chain key input The introduction of DSP digital signal processing dramatically changed the implementation of dynamics processors. In traditional analog designs, there is no practical means of "looking ahead" or statistically analyzing the content of a signal, instead requiring a function to respond to events that have already occurred. The supporting circuitry for filtering and dynamic control of attack and release is complex and expensive with limited accuracy.
The most significant advantages are the ability to analyze a signal before it is processed and statistically analyze recent history. These abilities allow a wide range of new topologies offering superior performance; some of which appear later in this note. The incremental cost of a single function implemented in DSP is very small, resulting in significant cost reduction when requiring multiple functions. Digital signal processing offers both greater accuracy and reduced cost.
Chapter 2 -- Basic Compressors Compressors reduce compress the dynamic range of the signal passing through them; they turn down the loudest signals dynamically.
A compressor begins turning down the signal by an amount set by the ratio control when the input signal exceeds the level set by the threshold control.
A compressor changes the dynamics for purposes of aesthetics, intelligibility, recording or broadcast limitations. For instance, an input dynamic range of dB might pass through a compressor and exit with a new dynamic range of 70 dB.
This clever bit of skullduggery is done in analog designs using a VCA whose gain is determined by a control voltage derived from the input signal other schemes exist but VCAs dominate. Digital designs use complex mathematical algorithms optimized for music and speech signal dynamics. Before compressors, a human did this at the mixing board and we called it gain riding.
The difficulty that sound systems have handling the full audio range dictates dynamic range reduction. If you turn it up as loud as you want for the average signals, then along come these huge musical peaks, which are vital to the punch and drama of the music, yet are way too large for the power amps and loudspeakers to handle.
Or going the other way, if you set the system gain to prevent these overload occurrences, then when things get nice and quiet, and the vocals drop real low, nobody can hear a thing. To fix this, you need a compressor. Using it is quite simple: Set a threshold point, above which everything will be turned down a certain amount and then select a ratio defining just how much a "certain amount" is, in dB.
All audio below the threshold point is unaffected and all audio above this point is compressed by the ratio amount. The earlier example of reducing dB to 70 dB requires a ratio setting of 1.
The key to understanding compressors is always to think in terms of increasing level changes in dB above the threshold point. A compressor makes these increases smaller. From the example, for every 1. In this regard, compressors make loud sounds quieter.
If the sound gets louder by 1. Broadband Compression Broadband compression is the simplest form of compression, where all frequencies are compressed equally and the side-chain is equally sensitive to all frequencies.
An rms detector is typically used and the basic gain computer side-chain controls are threshold, ratio, attack and release as shown in Figure 3a. The response of an above-threshold compressor with a threshold of dB and a ratio of is shown in Figure 3b. Figure 3a. Broadband compressor block diagram Figure 3b. Broadband compressor response graph Figure 3c. Halogen compressor block Compressor Uses Reduce the dynamic range of a vocal to enable it to remain present and audible in a mix when competing with other amplified instruments.
Used when mixing both live and recorded material. Reduce dynamic range of vocalists and other musical instruments that exceed the recording or reproduction capability. Prevent clipping and distortion in live sound systems or recording chains. Smooth and balance instruments such as bass guitars with wide dynamic range and large string-to-string level variations, or equalize different brass instrument volume levels.
Reduce sibilance de-essing. Produce louder recordings for broadcast. Even out paging loudness variations due to different announcing voices. Control the creation of sound. When used in conjunction with microphones and instrument pick-ups, compressors help determine the final timbre by selectively compressing specific frequencies and waveforms. Common examples are fattening drum sounds, increasing guitar sustain, vocal smoothing, and bringing up punching specific sounds in the mix.
Compression is available in several forms. As explained earlier the basics are the same for all types: a side-chain level is compared to a threshold and a gain computer uses the difference between the threshold and the side-chain level in combination with the side-chain control settings to determine the gain. Each of the compression techniques that follow, evolved to satisfy a specific need. Chapter 3 -- Side-Chain Controls A number of parameters govern side-chain activity, but the four primarily ones are Threshold, Ratio, Attack and Release.
Some dynamics controllers offer front-panel adjustment of these parameters, or software control, while others auto-set them at optimum values. Threshold and Ratio have unambiguous definitions: Threshold Like crossing through a doorway, this is the beginning point of gain adjustment. When the input signal is below the threshold for compressors, or above the threshold for expanders, a dynamics processor acts like a piece of wire.
Above the threshold, the side-chain asserts itself and reduces the volume or the other way around for an expander. A good expander extends the range to dBu for low-level signals. Ratio Once the signal exceeds the threshold setting, just how much the volume changes depends on the ratio setting.
Computer Music Tutorial
A straight wire has a ratio of -- the output tracks the input -- a 2 dB change at the input produces a 2 dB change at the output. A severe ratio is For a ratio, a 10 dB blast at the input changes only 1 dB at the output -- this represents heavy processing.
Kinder, gentler ratios are in the to range. Figure 4 shows the normal ranges for ratio controls of to If provided, the lower limit of is for bypassing. Figure 4. Input vs.
Gain can be done in the main signal path, or in the side-chain as control offset. Hard Times Unfortunately precise definitions for the terms attack and release do not exist due to a lack of industry standards. Moreover, manufacturers make this worst by not explaining how they define the terms. Most don't, they just list a range of settings, leaving the user to guess if the time shown represents how long it takes to get to the end of the gain change, or to the middle, or the 3-dB point, or what -- caveat emptor.
Attack Defines how quickly the function responds to an increase in side-chain input level above the threshold. For compressors and AGC, this defines how quickly the gain is turned down. For gates and expanders, this defines how quickly the gain is turned up. Because increasing time has a diminishing effect on gain for compressors, it is practical to specify attack as the time required for gain to settle to a defined percent of final value.
Attack times for compressors generally range between 25 ms and ms. In expand mode, attack time determines the rate of gain increase as the control signal moves toward or above the set threshold. In gate mode, attack time determines how quickly the gate opens once the control signal exceeds the threshold setting.
In ducker mode, attack time determines how quickly the signal is reduced as the control signal exceeds the threshold setting. Release Defines how quickly the function responds to a decrease in side-chain input level below the threshold. For compressors and AGC, this defines how quickly the gain is turned back up once these processes have stopped. For gates and expanders, this defines how quickly the gain is turned down. Release is typically defined by an RC resistor-capacitor time constant in the log domain, resulting in a constant dB per second gain change at the output.
It is important to understand the difference between release rate -- as determined by this control -- and release time. There is no industry standard and different manufacturers define this control differently. Rane defines this control, in a compressor for example, as how long it takes for the gain to change by 10 dB, not how long it takes to return to unity gain no gain reduction. Typical compressor and expander release settings are between 25 ms and 2 seconds. In gate mode, the release time determines how quickly the gate closes as the control signal drops below the threshold setting.
In expand mode, the release time determines how quickly the signal is turned down as the control signal moves below the set threshold.
In duck mode, the release time determines how quickly the signal is ramped up when the control signal drops below the threshold setting. Knee Unique to compressors, this function controls the action at the threshold point. Hard knee does nothing until the signal exceeds the threshold point, and then applies full compression. Use of a soft knee significantly reduces distortion caused by abrupt transitions from unity gain to a compressed signal.
An accurate soft knee response is difficult to achieve using analog methods. Digital implementations allow locating the center of the knee exactly at the threshold with a mathematical function defining a smooth transitioning from unity gain to the specified ratio. Note in Figure 5 that a proper soft knee response does not alter the ultimate gain reduction achieved above the knee, which commonly occurs in analog designs.
A soft knee is defined by the "span. Soft knee begins applying a small amount of compression just before the threshold point is reached, continues increasing compression through the threshold point and beyond, finally applying full compression to the highest level signals.
Depending on the application and source material, soft knee settings sound more natural. However for maximum loudness before compression equipment protection for instance use hard knee settings. Figure 5.
Compressor adjustable knee characteristics Chapter 4 -- Specialized Compressors This chapter explores very useful variations on the basic compressor technology. Adding a parametric equalizer section in the side-chain creates a frequency sensitive compressor; using a crossover allows split-band compression; putting a tracking filter into the main signal path and side-chain gives you a dynamic EQ; comparing broadband and bandpass energies produces a relative threshold dynamic EQ, which makes a terrific de-esser; while other clever additions solve the problems of automatic gain control and peak limiting.
Here are the details: Frequency Sensitive Compression Frequency sensitive compression is broadband compression as described above with the addition of side-chain equalization to make the detector more or less sensitive to certain frequencies. The basic topology is shown in Figure 6. Side-chain equalization may take the form of a parametric filter with variable boost, cut and bandwidth , high-cut filter, low-cut filter or all three.
In some cases, multiple parametric filters or a multiband graphic are used in the side-chain. If the amplitude of a frequency in the side-chain is reduced, the broadband compressor is less sensitive to it. If the amplitude of a frequency is boosted in the side-chain, the broadband compressor is more sensitive to it. Figure 6. Frequency sensitive compressor block diagram Split-Band Compression Split-band Compression divides the incoming signal into two or more frequency bands as shown in Figure 7.
Each band has its own side-chain detector and gain reduction is applied equally to all frequencies in the passband. After dynamics processing, the individual bands are re-combined into one signal.
The handling of in-band signals is the same as for the general-case compressor shown above. Figure 7. Split-band compressor block diagram This configuration is easily done with Rane's Halogen software and HAL processors as shown in Figure 8. This quickly expands into three, four or more frequency bands as required. Moreover, adding side-chain EQ and filters is just a drag and drop away.
Figure 8. The basic dynamic EQ uses a bandpass filter in the side-chain with variable center frequency and bandwidth. The side-chain detector is sensitive only to the passband frequencies. The basic topology is shown in Figure 9. Figure 9. The difference between the bandpass and broadband levels is compared to the threshold rather than the absolute rms value of the bandpass signal. The advantage of this type of dynamic EQ is that the relative amplitude of a band of frequencies, as compared to the broadband level, is maintained regardless of broadband amplitude.
The typical topology is shown in Figure Figure Relative threshold dynamic EQ block diagram De-essers De-essing limits or controls the sibilant content of speech.
Sibilance produces a hissing sound. English sibilant speech sounds are s , sh , z , or zh. De-essing is often confused as a type of dynamics processor. It's actually a specific application that is accomplished using many different types of dynamics processors. And contrary to popular belief, successful de-essing is not as simple as placing a bandpass or treble-boost filter in the side-chain and calling it done. True de-essing involves comparing the relative difference between the troublesome sibilants and the overall broadband signal, then setting a threshold based on this difference, therefore it is our experience that Relative Threshold Dynamic EQ as described above is the best dynamics processor for this task as it is able to maintain proper sibilant to non-sibilant balance regardless of level.
A good de-esser looks at the average level of the broadband signal 20 Hz to 20 kHz and compares it to the average level of a bandpass filter in the side-chain. The threshold setting defines the relative threshold, or difference, between broadband and bandpass levels, which result in compression of sibilants. Because de-essing depends on the ratio of sibilant to broadband signal levels, it is not affected by the absolute signal level, allowing the de-esser to maintain the correct ratio of broadband to sibilant material regardless of signal level, as shown in Figure Taming sibilance when the talker is quiet is just as important as when the talker is at a fevered pitch.
Figure 12 shows what happens using a primitive de-esser with a side-chain EQ. Sibilance during loud passages is attenuated, but there is no gain reduction during quiet passages, even though there may still be a significant amount of "sss" in the person's voice. For a given threshold, this often results in an overly aggressive effect during the loud choruses, and a completely ineffective result during the hissy, whispered verses.
Primitive de-esser with a simple side-chain. Varying input levels adversely affects de-essing. It is a circuit or algorithm that varies gain as a function of the input signal amplitude. Commonly found in pro audio applications where you want to automatically adjust the gain of different sound sources in order to maintain a constant loudness level at the output.
One of the most common applications is for speech. Not only do signal levels differ greatly between different source technologies but also between any two examples of the same technology, e. AGC is more similar to older compressor designs which compressed a signal about a threshold value see Appendix.
In these designs, gain was reduced for signals above the threshold and increased for signals below the threshold. One of the problems encountered with this type of compressor is the possibility of very high gain at low signal levels.
The Computer, Music, and Sound Models
A typical modern AGC implementation is shown in Figure 13a. Note that the traditional threshold control is now labeled target. The target is the desired, nominal level.
As with early compressors, gain is reduced for signals above the target and increased for signals below the target as shown in Figure 13b. The threshold is defined as the level below which the AGC circuit will not increase the gain. The implementation shown in Figure 13a indirectly determines the threshold for given maximum gain and ratio settings.
Other implementations exist, but the basics are the same. The hold parameter determines how long the above threshold gain is held in the absence of a signal. For speech, this feature allows the correct gain to be held during pauses. Figure 13a. AGC block diagram Figure 13b. AGC response graph Figure 13c. Halogen AGC control block Peak Limiter As expected, the basic topology of the limiter is similar to that of the compressor. Unlike the compressor, the limiter must ensure that a signal never exceeds the set threshold.
This requires the use of a peak responding detector and a fixed ratio of infinity Figure 14a illustrates the basic topology. The response of a limiter with a threshold of dB is shown in Figure 14b. Figure 14a. Peak limiter block diagram Figure 14b. Peak limiter response graph Figure 14c.
Halogen Limiter control block While the basic operation of a limiter is straightforward, coaxing one to sound good is challenging. Abrupt limiting causes significant alteration of the sound and determining the best release rate for a particular signal is problematic. Digital signal processing enables two additions to the basic topology which go a long way toward resolving these issues.
First, adding delay in the main signal path allows the side-chain to "see what's coming", and start to respond prior to the threshold actually being reached. The result is a softer leading edge resulting in a more natural sound. Second, looking at recent history gives the system knowledge of where the signal has been and where it is likely to go.
With this knowledge, the best release rate is set dynamically. Figure 15 illustrates the topology. Look-ahead peak limiter block diagram Primarily used for preventing equipment, media, and transmitter overloads, a peak limiter is to a compressor as a noise gate is to an expander more on this later. The most useful dynamics processor designs incorporate a separate peak limiter function independent of the compressor.
A separate peak limiter frees the compressor from the task of clamping wild excursions. The peak limiter plays level-police while the compressor persuades more gently. Peak Limiter Uses Prevent clipping and distortion in power amplifiers. Protection of loudspeakers from damage resulting from destructive transients like a dropped microphone. Preventing overs digital clipping during recording. Preventing overmodulation of the transmitted signal in broadcast. For example, a compressed input dynamic range of 70 dB might pass through an expander and exit with a new expanded dynamic range of dB.
As shown in Figure 16a, the topology for an expander looks just like a compressor. The difference is what the gain computer is directed to do with the difference between the threshold and the detected signal level. Unlike a compressor, the expander reduces gain for signals below the threshold. The ratio still defines output change verses input change as shown in Figure 16b.
In this example, the ratio is For every 10 dB of reduction in input signal, the output is reduced by 20 dB. Operating in this manner, they make the quiet parts quieter. A compressor keeps the loud parts from getting too loud, an expander makes the quiet parts quieter. Figure 16a. Expander block diagram Figure 16b.
Expander response graph Figure 16c. Halogen Expander block The term downward expander or downward expansion evolved to describe this type of application. The most common use is noise reduction. Musical applications of microprocessors. Google Scholar D. Deutsch, editor The Psychology of Music. Academic Press, New York.
Google Scholar B. Introduction to electroacoustic music. Prentice Hall. Google Scholar J. Pierce The Science of Musical Sound. Google Scholar R. Cogan New images of musical sound.
Harvard University Press. Google Scholar C. Roads, ed. Composers and the computer. Kaufman, Los Altos, Ca. Google Scholar Articles J. Strawn, ed. Digital audio signal processing. Google Scholar P. Manning Electronic and computer music. Clarendon Press, Oxford. Jerse Computer Music: Synthesis, Composition and Performance. Google Scholar Nuova Atlantide: il continente della musica elettronica , L.
Frequency Modulation: Theory and Applications. Yamaha Foundation, Morrill forthcoming. The little book of computer music instruments. Rabiner A unified approach to short-time Fourier analysis and synthesis. Arfib Digital synthesis of complex spectra by means of multiplication of non-linear distorted sine waves.
Audio Eng. Bacry, A. Zak Proof of the completeness of lattice states in the kqrepresentation. B12 , p. Bastiaans CrossRef Google Scholar C. Cadoz, A. Florens Responsive input devices and sound sybthesis by simulation of instrumental mechanisms: the Cordis system. CrossRef Google Scholar S. Robinson A comparison of orthogonal transformations for digital speech processing. CrossRef Google Scholar G. Charbonneau Timbre and the perceptual effect of three types of data reduction.
CrossRef Google Scholar J. Chowning The synthesis of complex audio spectra by means of frequency modulation. Google Scholar LO. A novel method of speech-sound analysis and synthesis.
Customers who viewed this item also viewed
Dissertation, Harvard Univ. Google Scholar M. Dolson The phase vocoder: a tutorial. CrossRef Google Scholar D. Gabor Theory of communication.
Gibson The senses considered as perceptual systems. Houghton Mifflin Company, Boston. Google Scholar G. Kott Moorer Perceptual evaluation of synthesized musical instrument tones. Morlet Decomposition of Hardy functions into square integrable wavelets of constant shape. SIAM J.
Representation and analysis of signals.
Part 1: the use of orthogonalized exponentials. John Hopkins Univ.
Google Scholar W. Hartmann The frequency-domain grating. Ruiz Synthesizing sound by solving the wave equation for vibrating objects.
Kronland-Martinet, J. Grossman Analysis of sound patterns through wavelet transforms. CrossRef Google Scholar M. Pomerantz, ed. Perceptual organization. Eribaum, Hillsdale, N. Cf in particular A. Shepard: Psychophysical complementarity. Steiglitz Synthesis of timbral families by warped linear prediction. Le Brun Digital waveshaping synthesis. Loy Musicians make a standard: the MIDI phenomenon.
Kohut Electronic simulation of violin resonances. Signal processing aspects of computer music: a survey. Proceedings of the IEEE 65, — The use of the phase vocoder in computer music applications.
Follow the author
About that reverberation business.Cogan Tim Stinchcombe wrote: Le Brun It has some great stuff about electromagnetic field generation when feeding current into opamps, giving some techniques to minimise generation of EM fields as well as some cancellation of EM pickup.
This idea was extended by Rodet and Depalle [ ] to include shaped amplitudes in the time domain. Gate response graph Figure 17c. A three-byte MIDI message requires nearly 1 millisecond for transmission. Kohut Typical settings for a bass guitar are a ratio of , with a fast attack of 25 ms and a slow release of around ms. Duckers A ducker is a dynamics processor that lowers ducks the level of one audio signal based upon the level of a second audio signal or a control trigger.