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Audio Processing

Audio processing covers many diverse fields, all involved in presenting sound to human listeners. Three areas are prominent: (1) high fidelity music reproduction, such as in audio compact discs, (2) voice telecommunications, another name for telephone networks, and (3) synthetic speech, where computers generate and recognize human voice patterns. While these applications have different goals and problems, they are linked by a common umpire: the human ear. Digital Signal Processing has produced revolutionary changes in these and other areas of audio processing.

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he most sensitive between 1 kHz and 4 kHz. S o ft e r Lo u d e r The primary advantage of having two ears is the ability to identify the direction of the sound. Human listeners can detect the difference between two sound sources that are placed as little as three degrees apart, about the width of a person at 10 meters. This directional information is obtained in two separate ways. First, frequencies above about 1 kHz are strongly shadowed by the head. In other words, the ear nearest the sound receives a stronger signal than the ear on the opposite side of the head. The second clue to directionality is that the ear on the far side of the head hears the sound slightly ater than the near ear, due to its greater distance from the source. Based on a typical head size (about 22 cm) and the speed of sound (about 340 meters per second), an angular discrimination of three degrees requires a timing precision of about 30 microseconds. Since this timing requires the volley principle, this clue to directionality is predominately used for sounds less than about 1 kHz. Both these sources of directional information are greatly aided by the ability to turn the head and observe the change in the signals. An interesting sensation occurs when a listener is presented with exactly the same sounds to both ears, such as listening to monaural sound through headphones. The brain concludes that the sound is coming from the center of the listener's head! While human hearing can determine the directiona sound is from, it does poorly in identifying the distance to the sound source. This is because there are few clues available in a sound wave that can provide this information. Human hearing weakly perceives that high frequency sounds are nearby, while low frequency sounds are distant. This is because sound waves dissipate their higher frequencies as they propagate long distances. Echo content is another weak clue to distance, providing a perception of the room size. For example, Chapter 22- Audio Processing 355 Time (milliseconds) 0 1 2 3 4 5 -2 -1 0 1 2 3 b. 1 kHz - 3 kHz sine waves Time (milliseconds) 0 1 2 3 4 5 -2 -1 0 1 2 3 a. 1 kHz + 3 kHz sine waves FIGURE 22-2 Phase detection of the human ear. The human ear is very insensitive to the relative phase of the component sinusoids. For example, these two waveforms would sound identical, because the mplitudes of ir components are the same, even though their relative phases re different. A m p lit u d e A m p lit u d e sounds in a large auditorium will contain echoes at about 100 millisecond intervals, while 10 milliseconds is typical for a small office. Some species have solved this ranging problem by using active sonar. For example, bats and dolphins produce clicks and squeaks that reflect from nearby objects. By measuring the interval between transmission and echo, these animals can locate objects with about 1 cm resolution. Experiments have shown that some humans, particularly the blind, can also use active echo localization to a small extent. Timbre The perception of a continuous sound, such as a note from a musical instrument, is often divided into three parts: loudness, pitch, and timbre (pronounced "timber"). Loudness is a measure of sound wave intensity, as previously described. Pitch is the frequency of the fundamental component in the sound, that is, the frequency with which the waveform repeats itself. While there are subtle effects in both these perceptions, they are a straightforward match with easily characterized physical quantities. Timbre is more complicated, being determined by the harmonic content of the signal. Figure 22-2 illustrates two waveforms, each formed by adding a 1 kHz sine wave with an amplitude of one, to a 3 kHz sine wave with an amplitude of one-half. The difference between the two waveforms is that the one shown in (b) has the higher frequency inverted before the addition. Put another way, the third harmonic (3 kHz) is phase shifted by 180 degrees compared to the first harmonic (1 kHz). In spite of the very different time domain waveforms, these two signals sound identical. This is because hearing is based on the amplitude of the frequencies, and is very insensitive to their phase. The shape of the time domain waveform is only indirectly related to hearing, and usually not considered in audio systems. The Scientist and Engineer's Guide to Digital Signal Processing356 The ear's insensitivity to phase can be understood by examining how sound propagates through the environment. Suppose you are listening to a person speaking across a small room. Much of the sound reaching your ears is reflected from the walls, ceiling and floor. Since sound propagation depends on frequency (such as: attenuation, reflection, and resonance), different frequencies will reach your ear through different paths. This means that the relative phase of each frequency will change as you move about the room. Since the ear disregards these phase variations, you perceive the voice as unchanging as you move position. From a physics standpoint, the phase of an audio signal becomes randomized as it propagates through a complex environment. Put another way, the ear is insensitive to phase because it contains little useful information. However, it cannot be said that the ear is completely deaf to the phase. This is because a phase change can rearrange the time sequence of an audio signal. An example is the chirp system (Chapter 11) that changes an impulse into a much longer duration signal. Although they differ only in their phase, the ear can distinguish between the two sounds because of their difference in duration. For the most part, this is just a curiosity, not something that happens in the normal listening environment. Suppose that we ask a violinist to play a note, say, the A below middle C. When the waveform is displayed on an oscilloscope, it appear much as the sawtooth shown in Fig. 22-3a. This is a result of the sticky rosin applied to the fibers of the violinist's bow. As the bow is drawn across the string, the waveform is formed as the string sticks to the bow, is pulled back, and eventually breaks free. This cycle repeats itself over and over resulting in the sawtooth waveform. Figure 22-3b shows how this sound is perceived by the ear, a frequency of 220 hertz, plus harmonics at 440, 660, 880 hertz, etc. If this note were played on another instrument, the waveform would look different; however, the ear would still hear a frequency of 220 hertz plus the harmonics. Since the two instruments produce the same fundamental frequency for this note, they sound similar, and are said to have identical pitch. Since the relative amplitude of the harmonics is different, they will not sound identical, and will be said to have different timbre. It is often said that timbre is determined by the shape of the waveform. This is true, but slightly misleading. The perception of timbre results from the ear detecting harmonics. While harmonic content is determined by the shape of the waveform, the insensitivity of the ear to phase makes the relationship very one- sided. That is, a particular waveform will have only one timbre, while a particular timbre has an infinite number of possible waveforms. The ear is very accustomed to hearing a fundamental plus harmonics. If a listener is presented with the combination of a 1 kHz and 3 kHz sine wave, they will report that it sounds natural and pleasant. If sine waves of 1 kHz and 3.1 kHz are used, it will sound objectionable. Chapter 22- Audio Processing 357 Frequency (hertz) 0 200 400 600 800 1000 1200 1400 1600 0 1 2 3 4 b. Frequency spectrum harmonics fundamental Time (milliseconds) 0 5 10 15 20 25 -8 -4 0 4 8 a. Time domain waveform FIGURE 22-3 Violin waveform. A bowed violin produces a sawtooth waveform, as illustrated in (a). The sound heard by the ear is shown in (b), the fundamental frequency plus harmonics. A m p lit u d e A m p lit u d e B C D E F G B C D E F G B C D E F G B C D E F G B C D E F G B C D E F G B C D E F G B C A- 27.5 Hz A- 55 Hz A- 110 Hz A- 880 Hz A- 1760 Hz A- 3520 HzA- 440 HzA- 220 Hz C- 262 Hz (Middle C) FIGURE 22-4 The Piano keyboard. The keyboard of the piano is a logarithmicfrequency scale, with the fundamental frequency doubling after every seven white keys. These white keys are the notes: A, B, C, D, E, F and G. This is the basis of the standard musical scale, as illustrated by the piano keyboard in Fig. 22-4. Striking the farthest left key on the piano produces a fundamental frequency of 27.5 hertz, plus harmonics at 55, 110, 220, 440, 880 hertz, etc. (there are also harmonics between these frequencies, but they aren't important for this discussion). These harmonics correspond to the fundamental frequency produced by other keys on the keyboard. Specifically, every seventh white key is a harmonic of the far left key. That is, the eighth key from the left has a fundamental frequency of 55 hertz, the 15th key has a fundamental frequency of 110 hertz, etc. Being harmonics of each other, these keys sound similar when played, and are harmonious when played in unison. For this reason, they are ll called the note, A. In this same manner, the white key immediate right of each A is called a B, and they are all harmonics of each other. This pattern repeats for the seven notes: A, B, C, D, E, F, and G. The term octave means a factor of two in frequency. On the piano, one octave comprises eight white keys, accounting for the name (octo is Latin for eight). In other words, the piano’s frequency doubles after every seven white keys, and the entire keyboard spans a little over seven octaves. The range of human hearing is generally quoted as 20 hertz to 20 kHz, The Scientist and Engineer's Guide to Digital Signal Processing358 corresponding to about ½ octave to the left, and two octaves to the right of the piano keyboard. Since octaves are based on doubling the frequency every fixed number of keys, they are a logarithmic representation of frequency. This is important because audio information is generally distributed in this same way. For example, as much audio information is carried in the octave between 50 hertz and 100 hertz, as in the octave between 10 kHz and 20 kHz. Even though the piano only covers about 20% of the frequencies that humans can hear (4 kHz out of 20 kHz), it can produce more than 70% of the audio information that humans can perceive (7 out of 10 octaves). Likewise, the highest frequency a human can detect drops from about 20 kHz to 10 kHz over the course of an adult's lifetime. However, this is only a loss of about 10% of the hearing ability (one octave out of ten). As shown next, this logarithmic distribution of information directly affects the required sampling rate of audio signals. Sound Quality vs. Data Rate When designing a digital audio system there are two questions that need to be asked: (1) how good does it need to sound? and (2) what data rate can be tolerated? The answer to these questions usually results in one of three categories. First, high fidelity music, where sound quality is of the greatest importance, and almost any data rate will be acceptable. Second, telephone communication, requiring natural sounding speech and a low data rate to reduce the system cost. Third, compressed speech, where reducing the data rate is very important and some unnaturalness in the sound quality can be tolerated. This includes military communication, cellular telephones, and digitally stored speech for voice mail and multimedia. Table 22-2 shows the tradeoff between sound quality and data rate for these three categories. High fidelity music systems sample fast enough (44.1 kHz), and with enough precision (16 bits), that they can capture virtually all of the sounds that humans are capable of hearing. This magnificent sound quality comes at the price of a high data rate, 44.1 kHz × 16 bits = 706k bits/sec. This is pure brute force. Whereas music requires a bandwidth of 20 kHz, natural sounding speech only requires about 3.2 kHz. Even though the frequency range has been reduced to only 16% (3.2 kHz out of 20 kHz), the signal still contains 80% of the original sound information (8 out of 10 octaves). Telecommunication systems typically operate with a sampling rate of about 8 kHz, allowing natural sounding speech, but greatly reduced music quality. You are probably already familiar with this difference in sound quality: FM radio stations broadcast with a bandwidth of almost 20 kHz, while AM radio stations are limited to about 3.2 kHz. Voices sound normal on the AM stations, but the music is weak and unsatisfying. Voice-only systems also reduce the precision from 16 bits to 12 bits per sample, with little noticeable change in the sound quality. This can be reduced to only 8 bits per sample if the quantization step size is made unequal. This is a widespread procedure called companding, and will be Chapter 22- Audio Processing 359 TABLE 22-2 Audio data rate vs. sound quality. The sound quality of a digitized audio signal depends on its d ta rat, the product of its sampling rate and number of bits per sample. This can be broken into three categories, high fidelity music (706 kbits/sec), telephone quality speech (64 kbits/sec), and compressed speech (4 kbits/sec). Sound Quality Required Bandwidth Sampling rate Number of bits Data rate (bits/sec) Comments High fidelity music (compact disc) 5 Hz to 20 kHz 44.1 kHz 16 bit 706k Satisfies even the most picky audiophile. Better than human hearing. Telephone quality speech200 Hz to 3.2 kHz 8 kHz 12 bit 96k Good speech quality, but very poor for music. (with companding) 200 Hz to 3.2 kHz 8 kHz 8 bit 64k Nonlinear ADC reduces the data rate by 50%. A very common technique. Speech encoded by Linear Predictive Coding 200 Hz to 3.2 kHz 8 kHz 12 bit 4k DSP speech compression technique. Very low data rates, poor voice quality. discussed later in this chapter. An 8 kHz sampling rate, with an ADC precision of 8 bits per sample, results in a data rate of 64k bits/sec. This is the brute force data rate for natural sounding speech. Notice that speech requires less than 10% of the data rate of high fidelity music. The data rate of 64k bits/sec represents the straightforward application of sampling and quantization theory to audio signals. Techniques for lowering the data rate further are based on compressing the data stream by removing the inherent redundancies in speech signals. Data compression is the topic of Chapter 27. One of the most efficient ways of compressing an audio signal is Linear Predictive Coding (LPC), of which there are several variations and subgroups. Depending on the speech quality required, LPC can reduce the data rate to as little as 2-6k bits/sec. We will revisit LPC later in this chapter with speech synthesis. High Fidelity Audio Audiophiles demand the utmost sound quality, and all other factors are treated as secondary. If you had to describe the mindset in one word, it would be: overkill. Rather than just matching the abilities of the human ear, these systems are designed to exceed the limits of hearing. It's the only way to be sure that the reproduced music is pristine. Digital audio was brought to the world by the compact laser disc, or CD. This was a revolution in music; the sound quality of the CD system far exceeds older systems, such as records and tapes. DSP has been at the forefront of this technology. The Scientist and Engineer's Guide to Digital Signal Processing360 0.5 µm pit width 1.6 µm track spacing 0.8 µm minimum length3.5 µm maximum length direction readout FIGURE 22-5 Compact disc surface. Micron size pits are burned into the surface of the CD to represent ones and zeros. This results in a data density of 1 bit per µm2, or one million bits per mm2. The pit depth is 0.16 µm. Figure 22-5 illustrates the surface of a compact laser disc, such as viewed through a high power microscope. The main surface is shiny (reflective of light), with the digital information stored as a series of dark pits burned on the surface with a laser. The information is arranged in a single track that spirals from the inside to the outside, the opposite of a phonograph record. The rotation of the CD is changed from about 210 to 480 rpm as the information is read from the outside to the inside of the spiral, making the scanning velocity a constant 1.2 meters per second. (In comparison, phonograph records spin at a fixed rate, such as 33, 45 or 78 rpm). During playback, an optical sensor detects if the surface is reflective or nonreflective, generating the corresponding binary information. As shown by the geometry in Fig. 22-5, the CD stores about 1 bit per (µm)2, corresponding to 1 million bits per (mm)2, and 15 billion bits per disk. This is about the same feature size used in integrated circuit manufacturing, and for a good reason. One of the properties of light is that it cannot be focused to smaller than about one-half wavelength, or 0.3 µm. Since both integrated circuits and laser disks are created by optical means, the fuzziness of light below 0.3 µm limits how small of features can be used. Figure 22-6 shows a block diagram of a typical compact disc playback system. The raw data rate is 4.3 million bits per second, corresponding to 1 bit each 0.28 µm of track length. However, this is in conflict with the specified geometry of the CD; each pit must be no shorter than 0.8 µm, and no longer than 3.5 µm. In other words, each binary one must be part of a group of 3 to 13 ones. This has the advantage of reducing the error rate due to the optical pickup, but how do you force the binary data to comply with this strange bunching? The answer is an encoding scheme called eight-to-fourteen modulation (EFM). Instead of directly storing a byte of data on the disc, the 8 bits are passed through a look-up table that pops out 14 bits. These 14 bits have the desired bunching characteristics, and are stored on the laser disc. Upon playback, the binary values read from the disc are passed through the inverse of the EFM look-up table, resulting in each 14 bit group being turned back into the correct 8 bits. Chapter 22- Audio Processing 361 Reed- Solomon decoding Sample rate converter (×4) 14 bit DAC Bessel Filter Power Amplifier EFM decoding Optical pickup Speaker Sample rate converter (×4) 14 bit DAC Bessel Filter Power Amplifier Speaker (4.3 Mbits/sec) serial data (706 Kbits/sec) 16 bit samples at 44.1 kHz 14 bit samplesat 176.4 kHz Compact disc Left channel Right channel FIGURE 22-6 Compact disc playback block diagram. The digital information is retrieved from the disc with an optical sensor, corrected for EFM and Reed-Solomon encoding, and converted to stereo analog signals. In addition to EFM, the data are encoded in a format called two-level Reed- Solomon coding. This involves combining the left and right stereo channels along with data for error detection and correction. Digital errors detected during playback are either: co rected by using the redundant data in the encoding scheme, concealed by interpolating between adjacent samples, or muted by setting the sample value to zero. These encoding schemes result in the data rate being tripled, i.e., 1.4 Mbits/sec for the stereo audio signals versus 4.3 Mbits/sec stored on the disc. After decoding and error correction, the audio signals are represented as 16 bit samples at a 44.1 kHz sampling rate. In the simplest system, these signals could be run through a 16 bit DAC, followed by a low-pass analog filter. However, this would require high performance analog electronics to pass frequencies below 20 kHz, while rejecting all frequencies above 22.05 kHz, ½ of the sampling rate. A more common method is to use a multirat technique, that is, convert the digital data to a higher sampling rate before the DAC. A factor of four is commonly used, converting from 44.1 kHz to 176.4 kHz. This is called interpolation, and can be explained as a two step process (although it may not actually be carried out this way). First, three samples with a value of zero are placed between the original samples, producing the higher sampling rate. In the frequency domain, this has the effect of duplicating the 0 to 22.05 kHz spectrum three times, at 22.05 to 44.1 kHz, 41 to 66.15 kHz, and 66.15 to 88.2 kHz. In the second step, an efficient digital filter is used to remove the newly added frequencies. The sample rate increase makes the sampling interval smaller, resulting in a smoother signal being generated by the DAC. The signal still contains frequencies between 20 Hz and 20 kHz; however, the Nyquist frequency has been increased by a factor of four. This means that the analog filter only needs to pass frequencies below 20 kHz, while blocking frequencies above 88.2 kHz. This is usually done with a three pole Bessel filter. Why use a Bessel filter if the ear is insensitive to phase? Overkill, remember? The Scientist and Engineer's Guide to Digital Signal Processing362 Since there are four times as many samples, the number of bits per sample can be reduced from 16 bits to 15 bits, without degrading the sound quality. The sin(x)/x correction needed to compensate for the zeroth order hold of the DAC can be part of either the analog or digital filter. Audio systems with more than one channel are said to be in stereo (from the Greek word for solid, or three-dimensional). Multiple channels send sound to the listener from different directions, providing a more accurate reproduction of the original music. Music played through a monaural (one channel) system often sounds artificial and bland. In comparison, a good stereo reproduction makes the listener feel as if the musicians are only a few feet away. Since the 1960s, high fidelity music has used two channels (left and right), while motion pictures have used four channels (left, right, center, and surround). In early stereo recordings (say, the Beatles or the Mamas And The Papas), individual singers can often be heard in only one channel or the other. This rapidly progressed into a more sophisticated mix-down, where the sound from many microphones in the recording studio is combined into the two channels. Mix-down is an art, aimed at providing the listener with the perception of being there. The four channel sound used in motion pictures is called Dolby Stereo, with the home version called Dolby Surround Pro Logic. ("Dolby" and "Pro Logic" are trademarks of Dolby Laboratories Licensing Corp.). The four channels are encoded into the standard left and right channels, allowing regular two-channel stereo systems to reproduce the music. A Dolby decoder is used during playback to recreate the four channels of sound. The left and right channels, from speakers placed on each side of the movie or television screen, is similar to that of a regular two-channel stereo system. The speaker for the center channel is usually placed directly above or below the screen. Its purpose is to reproduce speech and other visually connected sounds, keeping them firmly centered on the screen, regardless of the seating position of the viewer/listener. The surround speakers are placed to the left and right of the listener, and may involve as many as twenty speakers in a large auditorium. The surround channel only contains midrange frequencies (say, 100 Hz to 7 kHz), and is del yed by 15 to 30 milliseconds. This delay makes the listener perceive that speech is coming from the screen, and not the sides. That is, the listener hears the speech coming from the front, followed by a delayed version of the speech coming from the sides. The listener's mind interprets the delayed signal as a reflection from the walls, and ignores it. Companding The data rate is important in telecommunication because it is directly proportional to the costof transmitting the signal. Saving bits is the same as saving money. Companding is a common technique for reducing the data rate of audio signals by making the quantization levels u equal. As previously mentioned, the loudest sound that can be tolerated (120 dB SPL) is about one- million times the amplitude of the weakest sound that can be detected (0 dB Chapter 22- Audio Processing 363 EQUATION 22-1 Mu law companding. This equation provides the nonlinearity for µ255 law companding. The constant, µ, has a value of 255, accounting for the name of this standard. y ' ln(1%µx) ln(1%µ) for 0 # x # 1 y ' 1% ln(Ax) 1% ln(A) for 1/A # x # 1 for 0 # x # 1/Ay ' Ax 1% ln(A) EQUATION 22-2 "A" law companding. The constant, A, has a value of 87.6. SPL). However, the ear cannot distinguish between sounds that are closer than about 1 dB (12% in amplitude) apart. In other words, there are only about 120 different loudness levels that can be detected, spaced logarithmically over an amplitude range of one-million. This is important for digitizing audio signals. If the quantization levels are equally spaced, 12 bits must be used to obtain telephone quality speech. However, only 8 bits are required if the quantization levels are made unequal, matching the characteristics of human hearing. This is quite intuitive: if the signal is small, the levels need to be very close together; if the signal is large, a larger spacing can be used. Companding can be carried out in three ways: (1) run the analog signal through a nonlinear circuit before reaching a linear 8 bit ADC, (2) use an 8 bit ADC that internally has unequally spaced steps, or (3) use a linear 12 bit ADC followed by a digital look-up table (12 bits in, 8 bits out). Each of these three options requires the same nonlinearity, just in a different place: an analog circuit, an ADC, or a digital circuit. Two nearly identical standards are used for companding curves: µ255 law (al o called mu law), used in North America, and "A" law, used in Europe. Both use a logarithmic nonlinearity, since this is what converts the spacing detectable by the human ear into a linear spacing. In equation form, the curves used in µ255 law and "A" law are given by: Figure 22-7 graphs these equations for the input variable, x, being b tween -1 and +1, resulting in the output variable also assuming values between -1 and +1. Equations 22-1 and 22-2 only handle positive input values; portions of the curves for negative input values are found from symmetry. As shown in (a), the curves for µ255 law and "A" law are nearly identical. The only significant difference is near the origin, shown in (b), where µ255 law is a smooth curve, and "A" law switches to a straight line. Producing a stable nonlinearity is a difficult task for analog electronics. One method is to use the logarithmic relationship between current and The Scientist and Engineer's Guide to Digital Signal Processing364 Input -1.0 0.0 1.0 -1.0 0.0 1.0 Input -0.05 0.00 0.05 -0.5 0.0 0.5 a. µ law and A law b. Zoom of zero crossing µ law A law O ut pu t FIGURE 22-7 Companding curves. The µ255 law and "A" law companding curves are nearly identical, differing only near the origin. Companding increases the amplitude when the signal is small, and decreases it when it is large. O ut pu t voltage across a pn diode junction, and then add circuitry to correct for the ghastly temperature drift. Most companding circuits take another strategy: approximate the nonlinearity with a group of straight lines. A typical scheme is to approximate the logarithmic curve with a group of 16 straight segments, called cords. The first bit of the 8 bit output indicates if the input is positive or negative. The next three bits identify which of the 8 positive or 8 negative cords is used. The last four bits break each cord into 16 equally spaced increments. As with most integrated circuits, companding chips have sophisticated and proprietary internal designs. Rather than worrying about what goes on inside of the chip, pay the most attention to the pinout and the specification sheet. Speech Synthesis and Recognition Computer generation and recognition of speech are formidable problems; many approaches have been tried, with only mild success. This is an active area of DSP research, and will undoubtedly remain so for many years to come. You will be very disappointed if you are expecting this section to describe how to build speech synthesis and recognition circuits. Only a brief introduction to the typical approaches can be presented here. Before starting, it should be pointed out that most commercial products that produce human sounding speech do not synthesize it, but merely play back a digitally recorded segment from a human speaker. This approach has great sound quality, but it is limited to the prerecorded words and phrases. Nearly all techniques for speech synthesis and recognition are based on the model of human speech production shown in Fig. 22-8. Most human speech sounds can be classified as either voic d or fricative. Voiced sounds occur when air is forced from the lungs, through the vocal cords, and out of the mouth and/or nose. The vocal cords are two thin flaps of tissue stretched across Chapter 22- Audio Processing 365 Noise Generator Pulse train Generator Digital Filter unvoiced voiced pitch vocal tract response synthetic speech FIGURE 22-8 Human speech model. Over a short segment of time, about 2 to 40 milliseconds, speech can be modeled by three parameters: (1) the selection of either a periodic or a noise excitation, (2) the pitch of the periodic excitation, and (3) the coefficients of a recursive linear filter mimicking the vocal tract response. the air flow, just behind the Adam's apple. In response to varying muscle tension, the vocal cords vibrate at frequencies between 50 and 1000 Hz, resulting in periodic puffs of air being injected into the throat. Vowels are an example of voiced sounds. In Fig. 22-8, voiced sounds are represented by the pulse train generator, with the pitch (i.e., the fundamental frequency of the waveform) being an adjustable parameter. In comparison, fricative sounds originate as random noise, not from vibration of the vocal cords. This occurs when the air flow is nearly blocked by the tongue, lips, and/or teeth, resulting in air turbulence near the constriction. Fricative sounds include: s, f, sh, z, v, and th. In the model of Fig. 22-8, fricatives are represented by a noise generator. Both these sound sources are modified by the acoustic cavities formed from the tongue, lips, mouth, throat, and nasal passages. Since sound propagation through these structures is a linear process, it can be represented as a linear filter with an appropriately chosen impulse response. In most cases, a recursive filter is used in the model, with the recursion coefficients specifying the filter's characteristics. Because the acoustic cavities have dimensions of several centimeters, the frequency response is primarily a series of resonances in the kilohertz range. In the jargon of audio processing, these resonance peaks are called the formant frequencies. By changing the relative position of the tongue and lips, the formant frequencies can be changed in both frequency and amplitude. Figure 22-9 shows a common way to display speech signals, the voice spectrogram, or voiceprint. The audio signal is broken into short segments, The Scientist and Engineer's Guide to Digital Signal Processing366 say 2 to 40 milliseconds, and the FFT used to find the frequency spectrum of each segment. These spectra are placed side-by-side, and converted into a grayscale image (low amplitude becomes light, and high amplitude becomes dark). This provides a graphical way of observing how the frequency content of speech changes with time. The segment length is chosen as a tradeoff between frequency resolution (favored by longer segments) and ime resolution (favored by shorter segments). As demonstrated by the a in rain, voiced sounds have a periodic time domain waveform, shown in (a), and a frequency spectrum that is a series of regularly spaced harmonics, shown in (b). In comparison, the s in storm, shows that fricatives have a noisy time domain signal, as in (c), and a noisy spectrum, displayed in (d). These spectra also show the shaping by the formant frequencies for both sounds. Also notice that the time-frequency display of the word rain looks similar both times it is spoken. Over a short period, say 25 milliseconds, a speech signal can be approximated by specifying three parameters: (1) the selection of either a periodic or random noise excitation, (2) the frequency of the periodic wave (if used), and (3) the coefficients of the digital filter used to mimic the vocal tract response. Continuous speech can then be synthesized by continually updating these three parameters about 40 times a second. This approach was responsible for one the early commercial successes of DSP: the peak & Spell, a widely marketed electronic learning aid for children. The sound quality of this type of speech synthesis is poor, sounding very mechanical and not quite human. However, it requires a very low data rate, typically only a few kbits/sec. This is also the basis for the linear predictive coding (LPC ) method of speech compression. Digitally recorded human speech is broken into short segments, and each is characterized according to the three parameters of the model. This typically requires about a dozen bytes per segment, or 2 to 6 kbytes/sec. The segment information is transmitted or stored as needed, and then reconstructed with the speech synthesizer. Speech recognition algorithms take this a step further by trying to recognize patterns in the extracted parameters. This typically involves comparing the segment information with templates of previously stored sounds, in an attempt to identify the spoken words. The problem is, this method does not work very well. It is useful for some applications, but is far below the capabilities of human listeners. To understand why speech recognition is so difficult for computers, imagine someone unexpectedly speaking the following sentence: Larger run medical buy dogs fortunate almost when. Of course, you will not understand the meaning of this sentence, because it has none. More important, you will probably not even understand all of the individual words that were spoken. This is basic to the way that humans Chapter 22- Audio Processing 367 Time (seconds) 0 0.5 1 1.5 2 2.5 0 1 2 3 4 5 The rain bow was seen after the rain storm Frequency (kHz) 0 1 2 3 4 5 0 200 400 600 800 1000 d. Frequency domain: s storm Time (milliseconds) 0 2 4 6 8 10 12 14 16 18 20 22 1200 1400 1600 1800 2000 2200 2400 2600 2800 a. Time domain: a in rain Time (milliseconds) 0 2 4 6 8 10 12 14 16 18 20 22 2000 2020 2040 2060 2080 2100 c. Time domain: s in storm Frequency (kHz) 0 1 2 3 4 5 0 500 1000 1500 2000 2500 b. Frequency domain: i rain Fr e q u e n cy ( kH z) A m p lit u d e A m p lit u d e A m p lit u d e A m p lit u d e ( × 1 0 ) FIGURE 22-9 Voice spectrogram. The spectrogram of the phrase: "The inbow was seen after the rain storm." Figures (a) and (b) shows the time and frequency signals for the voiced a in rain. Figures (c) and (d) show the time and frequency signals for the fricative s in storm. The Scientist and Engineer's Guide to Digital Signal Processing368 perceive and understand speech. Words are recognized by their sounds, but also by the context of the sentence, and the exp ctations of the listener. For example, imagine hearing the two sentences: The child wore a spider ring on Halloween. He was an American spy during the war. Even if exactly the same sounds were produced to convey the underlined words, listeners hear the correct words for the context. From your accumulated knowledge about the world, you know that children don't wear secret agents, and people don't become spooky jewelry during wartime. This usually isn't a conscious act, but an inherent part of human hearing. Most speech recognition algorithms rely only on the sound of the individual words, and not on their context. They attempt to recognize words, but not to understand speech. T is places them at a tremendous disadvantage compared to human listeners. Three annoyances are common in speech recognition systems: (1) The recognized speech must have distinct pauses between the words. This eliminates the need for the algorithm to deal with phrases that sound alike, but are composed of different words (i.e., spider ring and spy during). This is slow and awkward for people accustomed to speaking in an overlapping flow. (2) The vocabulary is often limited to only a few hundred words. This means that the algorithm only has to search a limited set to find the best match. As the vocabulary is made larger, the recognition time and error rate both increase. (3) The algorithm must be trained on each speaker. This requires each person using the system to speak each word to be recognized, often needing to be repeated five to ten times. This personalized database greatly increases the accuracy of the word recognition, but it is inconvenient and time consuming. The prize for developing a successful speech recognition technology is enormous. Speech is the quickest and most efficient way for humans to communicate. Speech recognition has the potential of replacing writing, typing, keyboard entry, and the electronic control provided by switches and knobs. It just needs to work a little better to become accepted by the commercial marketplace. Progress in speech recognition will likely come from the areas of artificial intelligence and neural networks as much as through DSP itself. Don't think of this as a technical diffi u ty; think of it as a technical opportunity. Nonlinear Audio Processing Digital filtering can improve audio signals in many ways. For instance, Wiener filtering can be used to separate frequencies that are mainly signal, from frequencies that are mainly noise (see Chapter 17). Likewise, deconvolution can compensate for an undesired convolution, such as in the restoration of old Chapter 22- Audio Processing 369 Frequency 0 0.1 0.2 0.3 0.4 0.5 0 1 2 3 4 5 6 7 8 9 10 a. Signal spectrum Frequency 0 0.1 0.2 0.3 0.4 0.5 0 1 2 3 4 5 6 7 8 9 10 b. Noise spectrum FIGURE 22-10 Spectra of speech and noise. While the frequency spectra of speech and noise generally overlap, there is some separation if the signal segment is made short enough. Figure (a) illustrates the spectrum of a 16 millisecond speech segment, showing that many frequencies carry little speech information, in this particular segment. Figure (b) illustrates the spectrum of a random noise source; all the components have a small amplitude. (These graphs are not of real signals, but illustrations to show the noise reduction technique). A m p lit u d e A m p lit u d e recordings (also discussed in Chapter 17). These types of linear techniques are the backbone of DSP. Several nonlinear techniques are also useful for audio processing. Two will be briefly described here. The first nonlinear technique is used for reducing wideband noise in speech signals. This type of noise includes: magnetic tape hiss, electronic noise in analog circuits, wind blowing by microphones, cheering crowds, etc. Linear filtering is of little use, because the frequencies in the noise completely overlap the frequencies in the voice signal, both covering the range from 200 hertz to 3.2 kHz. How can two signals be separated when they overlap in both the time domain and the frequency domain? Here's how it is done. In a short segment of speech, the amplitude of the frequency components are greatly unequal. As an example, Fig. 22-10a illustrates the frequency spectrum of a 16 millisecond segment of speech (i.e., 128 samples at an 8 kHz sampling rate). Most of the signal is contained in a few large amplitude frequencies. In contrast, (b) illustrates the spectrum when only random noise is present; it is very irregular, but more uniformly distributed at a low amplitude. Now the key concept: if both signal and noise are present, the two can be partially separated by looking at the mplitude of each frequency. If the amplitude is large, it is probably mostly signal, and should therefore be retained. If the amplitude is small, it can be attributed to mostly noise, and should therefore be discarded, i.e., set to zero. Mid-size frequency components are adjusted in some smooth manner between the two extremes. Another way to view this technique is as a time v rying Wiener filter. As you recall, the frequency response of the Wiener filter passes frequencies that are mostly signal, and rejects frequencies that are mostly noise. This The Scientist and Engineer's Guide to Digital Signal Processing370 requires a knowledge of the signal and noise spectra befor hand, so that the filter's frequency response can be determined. This nonlinear technique uses the same idea, except that the Wiener filter's frequency response is recalculated for each segment, based on the spectrum of that segment. In other words, the filter's frequency response changes from segment-to-segment, as determined by the characteristics of the signal itself. One of the difficulties in implementing this (and other) nonlinear techniques is that the overlap-add method for filtering long signals is not valid. Since the frequency response changes, the time domain waveform of each segment will no longer align with the neighboring segments. This can be overcome by remembering that audio information is encoded in frequency patterns that change over time, and not in the shape of the time domain waveform. A typical approach is to divide the original time domain signal into over apping segments. After processing, a smooth window is applied to each of the over- lapping segments before they are recombined. This provides a smooth transition of the frequency spectrum from one segment to the next. The second nonlinear technique is called homomorphic signal processing. This term literally means: the same structure. Addition is not the only way that noise and interference can be combined with a signal of interest; multiplication and convolution are also common means of mixing signals together. If signals are combined in a nonlinear way (i.e., anything other than addition), they cannot be separated by linear filtering. Homomorphic techniques attempt to separate signals combined in a nonlinear way by making the problem become linear. That is, the problem is converted to the same structure as a linear system. For example, consider an audio signal transmitted via an AM radio wave. As atmospheric conditions change, the received amplitude of the signal increases and decreases, resulting in the loudness of the received audio signal slowly changing over time. This can be modeled as the audio signal, represented by , being multiplied by a slowly varying signal, that represents thea[ ] g[ ] changing gain. This problem is usually handled in an electronic circuit called an automatic gain control (AGC), but it can also be corrected with nonlinear DSP. As shown in Fig. 22-11, the input signal, , is passed through thea[ ]×g[ ] logarithm function. From the identity,log (x y) ' log x % log y, this results in two signals that are combined by addition, i.e., . In otherl ga[ ]% logg[ ] words, the logarithm is the homomorphic transform that turns the nonlinear problem of multiplication i to the linear problem of addition. Next, the added signals are separated by a conventional linear filter, that is, some frequencies are passed, while others are rejected. For the AGC, the gain signal, , will be composed of very low frequencies, far below theg[ ] 200 hertz to 3.2 kHz band of the voice signal. The logarithm of these signals will have more complicated spectra, but the idea is the same: a high-pass filter is used to eliminate the varying gain component from the signal. Chapter 22- Audio Processing 371 Logarithm Linear Filter Anti-Logarithm loga[ ]loga[ ]% logg[ ] a [ ]×g[ ] a [ ] FIGURE 22-11 Homomorphic separation of multiplied signals. Taking the logarithm of the input signal transforms components that are multiplied into components that are add d. These components can then be separated by linear filtering, and the effect of the logarithm undone. Fourier Transform Logarithm Linear Filter Anti- InverseFourier Transform Logarithm Homomorphic Transform Inverse Homomorphic Transform X [ ] logX [ ] logX [ ]% logY [ ] X [ ]×Y [ ] FIGURE 22-12 Homomorphic separation of convolved signals. Components that have been convol d are converted into components that are added by taking the Fourier transform followed by the logarithm. After linear filtering to separate the added components, the original steps are undone. x[ ]x[ ]t y[ ] In effect, is converted into . In the last step, theloga[ ]% logg[ ] loga[ ] logarithm is undone by using the exponential function (the anti-logarithm, or ), producing the desired output signal, e x a [ ]. Figure 22-12 shows a homomorphic system for separating signals that have been convolved. An application where this has proven useful is in removing echoes from audio signals. That is, the audio signal is convolved with an impulse response consisting of a delta function plus a shifted and scaled delta function. The homomorphic transform for convolution is composed of two stages, the Fourier transform, changing the convolution into a multiplication, followed by the logarithm, turning the multiplication into an addition. As before, the signals are then separated by linear filtering, and the homomorphic transform undone. An interesting twist in Fig. 22-12 is that the linear filtering is dealing with frequency domain signals in the same way that time domain signals are usually processed. In other words, the time and frequency domains have been swapped from their normal use. For example, if FFT convolution were used to carry out the linear filtering stage, the "spectra" being multiplied would be in the time domain. This role reversal has given birth to a strange jargon. For instance, cepstrum (a rearrangment of spectrum) is the Fourier transform of the logarithm of the Fourier transform. Likewise, there are long-p ss and short- pass filters, rather than low-pass and high-pass filters. Some authors even use the terms Quefrency Alanysis and liftering. The Scientist and Engineer's Guide to Digital Signal Processing372 Keep in mind that these are simplified descriptions of sophisticated DSP algorithms; homomorphic processing is filled with subtle details. For example, the logarithm must be able to handle both negative and positive values in the input signal, since this is a characteristic of audio signals. This requires the use of the complex logarithm, a more advanced concept than the logarithm used in everyday science and engineering. When the linear filtering is restricted to be a zero phase filter, the complex log is found by taking the simple logarithm of the absolute value of the signal. After passing through the zero phase filter, the sign of the original signal is reapplied to the filtered signal. Another problem is aliasing that occurs when the logarithm is taken. For example, imagine digitizing a continuous si e wave. In accordance with the sampling theorem, two or more samples per cycle is sufficient. Now consider digitizing the logarithm of this continuous sine wave. The sharp corners require many more samples per cycle to capture the waveform, i.e., to prevent aliasing. The required sampling rate can easily be 100 times as great after the log, as before. Further, it doesn't matter if the logarithm is applied to the continuous signal, or to its digital representation; the result is the same. Aliasing will result unless the sampling rate is high enough to capture the sharp corners produced by the nonlinearity. The result is that audio signals may need to be sampled at 100 kHz or more, instead of only the standard 8 kHz. Even if these details are handled, there is no guarantee that the linearized signals can be separated by the linear filter. This is because the spectra of the linearized signals can overlap, even if the spectra of the original signals do not. For instance, imagine adding two sine waves, one at 1 kHz, and one at 2 kHz. Since these signals do not overlap in the frequency domain, they can be completely separated by linear filtering. Now imagine that these two sine waves are multiplied. Using homomorphic processing, the log is taken of the combined signal, resulting in the log of one sine wave plus the log of the other sine wave. The problem is, the logarithm of a sine wave contains many harmonics. Since the harmonics from the two signals overlap, their complete separation is not possible. In spite of these obstacles, homomorphic processing teaches an important lesson: signals should be processed in a manner consist nt with how they are formed. Put another way, the first step in any DSP task is to understand how information is represented in the signals being processed.

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Rhode Island Resource Directory

A Guide to Services Associated with RI 2-1-1

Full Resource Name:

Boys and Girls Clubs of Newport County

95 Church Street
Newport, RI 02840

(401) 847-6927

The Boys & Girls Clubs of Newport County is dedicated to âShaping Great Futuresâ and that journey begins here. Providing a fun, safe environment, positive relationships with adults, and hope and opportunity for all who come through our doors, the Boys and Girls Clubs of Newport County work with children of all ages and serve their families to enable their success.
the Boys and Girls Club of Newport Country have three Locations

The Central Clubhouse/ Administration located at:
95 Church Street, Newport RI 02840

The Northend Clubhouse located at:
The Florence Gray Center, 1 York Street, Newport RI 02840

Camp Grosvenor located at:
11 Miner Road, Saunderstown RI 02874

Primary Services:

  • Boys/Girls Clubs  [Show Definition Details]

  • Camps  [Show Definition Details]

  • Child Care Centers  [Show Definition Details]

  • Child Care Provider Referrals  [Show Definition Details]

  • Drop In Centers  [Show Definition Details]

  • Extended Child Care  [Show Definition Details]

  • Summer Food Service Programs  [Show Definition Details]

General Information

Mon-Thu 6:30AM-8:00PM, Fri 6:30AM-6:30PM
Sat Open by Specific Programming

North End Clubhouse Mon-Fri 2:30PM-6:00PM
Camp Grosvenor June 25-Aug17 8:30AM-4:00PM

Intake Process:

Become a member, fill out enrollment/ financial aid form and return to Boys and Girls Club.

Program Fees:

Youth Membership Fee $30.
Childcare Rates: 5 Years Old: Before and After School $162, Before Care $81, After Care $122
6+ Years Old: Before and After $147, Before Care $73, After Care $110
Financial Aid available through DHS.

Free for military families
Adult Memberships: All inclusive $400/yr, Seniors/Students #60, Basic $275/yr, seniors/ students 250
Payment Plans Available.

All are welcome. Childcare for ages 5 through 15. North End Clubhouse open to ages 10-18. Camp Grosvenor ages 5-14.

Is Shelter?

Resource Groups

  • Login» to Manage Groups

    Create groups to save and organize individual resources.

    Login or create an account to begin.

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Order a Reading from Me

Order a Reading from Me
Please send relevant information to zannastarr@gmail.com.

Saturday, May 26, 2018

Pyramid Reading with The Spirit Within Tarot

This spread comes from the Complete Book of Tarot Spreads by Evelin Burger and Johannes Fieborg (Sterling Publishing Co., Inc.) I use this as the textbook for the Basic Tarot course I teach at _The Magickal Circle School_, and I highly recommend it as a resource.

The purpose of my reading today is to explore the potential and possibilities related to a business venture I am considering that uses my tarot and astrology knowledge and skills. Using the Spirit Within Tarot by Steven Bright (Schiffer Publishing), I pulled ten cards into the following layout:

* * * * * * * * *

1, 2, 3, 4 – Task

In order to perform this task well, I need to get plenty of rest and relaxation and avoid excessive worry. There will be times when I am dissatisfied or when I tend to see only the three “empty cups” rather than the fourth being offered to me. This task requires that I face my fears, acknowledge and accept all aspects of who I am in order to create a stronger, more successful “whole person.” My intuitive powers and inner guide can provide everything I need to perform this task successfully.

5, 6, 7 – Obstacles

There is an element of “the unknown” in this situation as I prepare to step into a new activity, and that can create anxiety, but this is the right time for what I am doing as long as I maintain a sense of balance and a practical approach, with both feet on the ground. I need to nurture relationships and partnerships involved in this situation in order to avoid any unpleasantness in that area.

8, 9 – Help/solution

It may be necessary for something to end in order for me to pursue this new path. I can determine what that is by using my mind to analyze and evaluate everything I am doing. I have the ability to make sound decisions about this. There may also be another person with the qualities of the Queen of Swords who is able and willing to help me.

10 – Outcome of efforts

If I follow the advice of the preceding cards, I will be able to offer levelheaded yet sensitive advice from a mature perspective, confident in my ability to counsel and share what I understand and know at a deep personal level. I will be “on top” and in control, taking charge of my own direction.

Sunday, May 13, 2018

Reading with the Ethereal Visions Tarot

In my review of the book Journaling the Tarot: a little book of big questions by Andy Matzner, I wrote:
"For me, I think the fun will come in following this procedure:
(1) Draw a card
(2) Draw a second card to indicate the number of the question I need to ask.
(3) Check the book. Focus on the question chosen.
(4) Pull as many additional cards as I like to help me answer that question."
(To read the entire review, click HERE.)

Let’s see how this idea might work. I am using the Ethereal Visions Illuminated Tarot Deck by Matt Hughes.

Step 1. The Card: FIVE OF CUPS

Interestingly, this same card came forward in a deck interview I did in April with this same deck. Click HERE. In that reading, the 5 of Cups represented an answer to the question: “What would you like me to understand about myself?” My interpretation at that time: “You need to stop thinking so much about what you have lost, where you have failed, and what 'might have been.' Look up. Turn around. Seize the goblets that are waiting to be filled.” All of that still holds.

Step 2. The Number of the Question: Knight of Pentacles = 12

This card was drawn primarily to provide a number, but I think it is worthwhile to look at the card meaning as well. Matt Hughes provides the following keywords for the Knight of Pentacles --
“Upright: prepared, efficient, ambitious, practical”; “Reversed: stagnant, lazy, careless, idle.”

It will be useful to look at how those words plus my own interpretation of this card could help me understand what the cards want me to know. I sometimes see the Knight of Pentacles as representing a determined, careful, direct approach. That’s all well and good, but this Knight can also be “stodgy” or single-minded in pursuit of an objective, with little ability or desire to look up and not only see what is going on around him, but attempt to understand it and to take it into account.

Step 3. Question 12 under the Five of Cups in Journaling the Tarot:
"Whom or what do you need to exorcise from your life? What would that ritual look like? How do you imagine you would feel afterwards?"

That looks like three questions to me, so I am drawing three cards to answer:

Whom or what do you need to exorcise from your life?

Keywords from the guidebook for the Ethereal Visions Illuminated Tarot Deck –

  • “Upright: prosperity, pleasure, motherly, security”
  • “Reversed: imbalance, reclusive, superficial, martyrdom”

Wow. I know exactly what this refers to. I’m a private person in many ways, and I don’t really want to broadcast my personal problems and issues all over the internet. Let me just say that the keywords – and my own personal way of interpreting the Queen of Pentacles – make a whole lot of sense. And also let me say that the fact that this is Mother's Day is also very appropriate.

What would that ritual look like?

Keywords –

  • “Upright: focus, decisions, discovery, planning”
  • “Reversed: overconfidence, stunted, undetermined, blocked”

Yes to all of the upright keywords, while hoping and trying to avoid the conditions suggested by the reversed keywords. I also notice how the man is standing alone at the top of a tower, looking down or out at a vast landscape of sea and shore. I know that the “traditional” interpretation of this card is usually about gazing out at the “landscape” with confidence and even pride. To me, however, in my current state of mind, the image suggests the possibility of feeling overwhelmed by the number of things or issues that need to be addressed, which of course means that focus and planning are absolutely critical to counter that sense of being overwhelmed.

How do you imagine you would feel afterwards?

Keywords –

  • “Upright: pain, grief, deception, rejection”
  • “Reversed: “forgiveness, recovery, clarity, introspection”

Yep. Spot on. The questions posed by Andy Matzner and the way the cards both directed me to the questions and answered the questions – all of it – spot on.

Wednesday, May 9, 2018

REVIEW: Journaling the Tarot: a little book of big questions

Journaling the Tarot: a little book of big questions
180 pages
ISBN: 1987796888
Sold by: Amazon Digital Services LLC

When I receive a book for review, my first step is usually to find a quiet place and time where I can read that book. Not so with Andy Matzner’s Journaling the Tarot. Instead, the very nature of the book invited me to jump in the “play” immediately. There would be time later if I wanted to read and digest each page (and I will).

I love that the book offers several individual pages containing relevant quotations, and when we come to the page titled “Instructions” it is blank except for the statement, “[this page intentionally left blank.]” Seriously, my Virgo ascendant immediately thought, “Oh dear, an error.” To which my Sagittarius sun replied, “No! It’s perfect! Don’t you get it?!?!”

Starting with The Fool (always the best place to start, I think), Matzner presents two pages of around 20-30 questions that each card of the Tarot might ask us (actually, let’s face it, they DO ask us). I confess (possibly with great embarrassment, possibly not) that I am not the journaling type. But if you are, this book will give you prompt after prompt for your journal entries as you delve into why that particular card speaks to you and what it is saying.

For me, I think the fun will come in following this procedure:
(1) Draw a card
(2) Draw a second card to indicate the number of the question I need to ask. (Because the questions number between 20 and 30 for each card, and the Tarot only goes up to 22, I need to figure out a method that doesn’t exclude questions numbered 23 and above.)
(3) Check the book. Focus on the question chosen.
(4) Pull as many additional cards as I like to help me answer that question.

Stay tuned – a blog entry using this approach is quickly approaching!

If you’re looking for a book you can use immediately, that also provides marvelous insight into the purpose and meaning of each Tarot card, get a copy of Journaling the Tarot by Andy Matzner. You won’t regret it.

Andy Matzner

Andy Matzner is gender therapist, coach, author, and educator with an office in Roanoke, VA . He is a licensed clinical social worker and psychotherapist who also teaches psychology at Virginia Western Community College and gender and women’s studies at hollins University. His published books include The Tarot Activity Book: A collection of creative and therapeutic ideas for the cards.

Of himself, he writes: “In the end, my mission is simple: To learn new things, share those things with others, and empower people to expand their horizons and fulfill their potential. That’s why I do what I do.” – Andy Matzner

To read more about Andy Matzner, click HERE.

PRODUCT DESCRIPTION from Author/Publisher

”If your tarot cards could speak, what questions would they ask?

Questions are powerful catalysts for introspection and self-growth. They can lead us in surprising directions. Or reconnect us to long-buried truths. It is important to take time in our lives every so often to check in with ourselves and meditate on the thoughts and feelings that well-thought-out questions can generate.

Usually we expect tarot cards to provide us with answers. In this book however, you will see that the cards themselves generate questions that you can use as prompts for journaling and discussion.”

  • The Fool: Who or what are you currently taking on faith?
  • The High Priestess: What don’t you know? What is your next step?
  • Temperance: With whom should you join forces? Why might that be a good idea?
  • The Moon: How are you missing the point?
  • Four of Wands: What should you be celebrating? How can you best do so?
  • Eight of Cups: What parts of your past must you explore and resolve in order to move forward?
  • Three of Swords: Whom will you disappoint? Why? Can this be avoided? If so, how?
  • Seven of Pentacles: How are you appreciating what you have?

This 8 x 5-1/4 inch paperback book does not contain illustrations or art. Its pages are direct and uncluttered, each featuring 20-30 bullet-pointed questions that might be asked by a particular Tarot card.

Paper quality and binding are up to standard. Use of white space and type font/style make the book easy to read and reference.

In accordance with the FTC Guidelines for blogging and endorsements, I hereby disclose that this product was provided by the publisher for free. Other than the occasional review copy, I receive no monetary or in-kind compensation for my reviews.  The substance of my reviews is not influenced by whether I do or do not receive a review copy.

Thursday, May 3, 2018

Elderly Parent Reading with the Ghost Tarot

I am dealing with some issues concerning my elderly (age 96) mother right now, so I thought I would see what the cards might be able to tell me. Theresa Reed provides a great spread for this purpose on her blog: https://www.thetarotlady.com/mothers-and-daughters-cant-we-all-just-get-along/

This is a simple 3-card spread with the positional definitions (1) Me, (2) Her, and (3) Advice. It may seem a bit odd, but I felt led to use my Ghost Tarot (Davide Corsi / Lo Scarabeo) for this reading. Here we go.

(1) Me: DEATH ("The door that is always open.”)

Wow. I’m not really sure what to do with this. I think it may refer to the idea that in many ways I am acting as an agent of major changes in my mother’s life. The fact that she broke her hip certainly is the cause of her being in a rehabilitation hospital and a nursing home for the past two months as she recovers, but she wants to return to her apartment and seems unwilling or uninterested in making changes to accommodate the risk of re-injuring or new injuries that would sideline her once and for all. You see, she still drives a car and feels that she “can take care of herself.” She is fiercely independent. She does not want to hear about how she might need someone to stay with her for a few days after she gets back in the apartment, someone to help make sure she takes the right medications at the right time, and so forth. I am one of the primary agents pushing changes that she does not want, and does not feel she needs.

(2) Her: THREE OF SWORDS (“The cold of the soul can perceive the warmth of the heart, even during a storm.”)

Hopefully this means that even though the suggestions and changes being introduced are painful for her, she can still understand (at some level) that it’s all because people care about her and want to keep her safe. I think the 3 of Swords is a pretty accurate representation of the way things feel to her right now. It is interesting that the man on the card is holding an umbrella over the woman-ghost, trying to protect her. That is what I am doing (at least in my own mind!) My mother, however, is the type of person who is determined to do what she wants to do, when she wants, the way she wants. She is determined to “take care of herself.” Though she be but little, she is most definitely fierce!

(3) AdviceTHE MAGICIAN (“The Alchemist’s Will”)

I often see this card as advising us to take charge of our own lives using the tools at our disposal, channeling the energy of the Universe down into our everyday lives. If this advice is intended for me, it could be suggesting that I simply do my best with what I have to work with, to be aware of all that I have available to work with, and to switch gears as needed from a logical approach, to an emotional one, to whatever method or means is best suited to specific situations. Is it possible that The Magician is also advising me to give my mother permission to do the same?

Two Major Arcana cards tell me that this situation is of great significance, and the way things are handled matters a great deal to my mother’s well being and my own.


text:
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text:
August 7, 2020

Lions scratch up win against Seagulls

November 29, 2016 Signal Contributor 0

The women’s basketball team fell 81-76 last Tuesday, Nov. 22, in a close match against Rutgers University-Camden, who are currently tied for first in the New Jersey Athletic Conference (NJAC). The game was the opening conference match-up for both schools. […]

Around the Dorm 11/30: Sports super villains, Super Bowl prospects and MLB free agents

November 29, 2016 Signal Editorial Staff 0

In this week’s edition of Around the Dorm, “Ref” David Weinberg asks our panel of three experts — Matt Ajaj, EJ Paras and Marc Trotochaud — three questions: Do you believe the Warriors should embrace their “super villain” role? Which team has the best chance to win the Super Bowl? Where will Yoenis Céspedes and Aroldis Chapman end up this offseason? […]

Watching sports brings families together on holidays

November 28, 2016 Michael Battista 0

America had a chance to relax with family, eat a ton of food and watch three NFL games all on Thursday, Nov. 24. The games were great — even though the Dallas Cowboys won — which got me thinking about every major sport and their connection to a holiday.

The NFL has Thanksgiving, the NBA has Christmas Day, college football — especially the Rose Bowl — usually has New Year’s if it isn’t a Sunday, the NHL is trying to carve a place into New Year’s Day as well with it’s Winter Classic game — once again so long as it isn’t on a Sunday and MLB has Independence Day. […]

Lions feast on competition during Thanksgiving break

November 28, 2016 George Tatoris 0

The College’s swimming and diving teams had more to digest than turkey and stuffing over Thanksgiving break.

The Lions withstood a grueling dual meet against the New York University (NYU) Violets the previous Saturday, Nov. 19. While the women’s team were routed, 223-71, by the eighth-ranked Violets, the men’s team slipped past them in the final event of the meet for a 151-143 victory. […]

Lions offense scorches in season debut

November 28, 2016 Miguel Gonzalez 0

The Lions were off to a good start after defeating the John Jay College Bloodhounds, 75-58, in their home opener on Tuesday, Nov. 15. The team followed with a 79-68 win against the State University of New York (SUNY) Geneseo Knights, before conceding a 68-74 loss to Pennsylvania State University-Harrisburg on Sunday, Nov. 20. […]

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Step ‘n Copy

$80.00 $185.00

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Limited Edition Giclee
Signed & Numbered By The Artist

Step and Copy was a sport carried out usually by older children and teenagers in outport Newfoundland. The objective was to successfully leap from one ice pan to another by briefly stepping on one or two smaller pans, while others following would try to do the same. More accomplished steppers could navigate as many as 8 or 10 pans, and not always in a straight line. However, even the best sometimes fell into the icy water and had to remove their boots, socks, and pants and beat them dry on a wharf post. After this procedure, the guilty party would hang around and delay going home, hopeful that Mother would not notice. She usually noticed.

The foreground of this image features Youngs’ and Greenhams’ stages, on the Southside, Twillingate,and looks across Shoal Tickle towards Burton’s Point, Northside. The sailing vessel lying derelict in the harbour is the Lester N. Peckford. As kids we would swing from her ropes and cables into the salt water.

Additional information

13.5" x 9" Artist Proof, 20" x 13" Limited, 20" x 13" Artist Proof


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TUNe Will Sizzle in September

The annual TUNe meeting will be held this year, September 9, 10, and 11th at the sizzling Ameristar Casino Resort and Spa in St. Charles, Missouri

The featured topic at this year’s TUNe meeting will be choosing and maintaining an effective workforce. TUNe members will lead a presentation on the best practices for finding and hiring topnotch employees. Included in this presentation is how to evaluate employees and how to terminate ineffective employees without costly missteps.

“Finding good employees is one of the biggest issues in our industry,” said TUNe president, Robbie Parnell, of Medcom in South Carolina. “Eliminating poor performers can be an even bigger challenge. We have some members who have good success in both areas and we are going to have them share their techniques. I think this is something that concerns us all.”

While the meeting will include all the important new product information from Telescan, the star of the conference will undoubtedly be the fabulous, AAA four-diamond Ameristar Casino Resort (http://www.ameristar.com/St_CharlesGaming.aspx.). Located just north of St. Louis on the banks of the mighty Missouri River, this resort has something for everyone. In addition to the glamorous casino, it includes no less than eight dining choices from casual to elegant. If you need a little tension release you could try a dip in the indoor/outdoor pool, with poolside service, or try a hot rocks massage in their Ara Spa. Should you have time for more excitement, walk outside and you can experience the cobblestone streets of historic Old Towne St. Charles where Lewis and Clark met to start their epic journey. The historic district includes no less than a dozen world class restaurants, museums and unique shopping experiences at every turn.

“We’ve tried several different venues, but this is the best,” said Robbie. “The luxurious 800 sq. ft. suites alone make the trip worthwhile. There is so much to do you can have a complete vacation without leaving the resort.”

Don’t miss this year’s TUNe adventure. Make your reservation today. Contact Dan L’Heureux at dan@callconsult.net to register.

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FLIR K33 & K53 Thermal Imaging Camera

FLIR K33 K53 TICs Thermal Imaging Cameras for Firefighters

Streamlined High Performance Made Even More Economical

FLIR now gives you two new lower-cost, easier-to-use options without sacrificing the reliable clarity and performance that FLIR K-Series TICs are well-known for. The K33 and K53 feature FLIR’s FSX™ Flexible Scene Enhancement that intensifies structural and textural details in thermal images for better perspective and orientation to help improve situational awareness and give firefighters a greater sense of confidence and safety.

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FLIR K45, K55 and K65 Thermal imaging cameras

K Series Now with FSX

FSX enhances K45, K55 and K65 thermal images through real-time digital processing inside the camera. The result is an ultra-sharp image that shows extraordinary structural, edge, and other instantly-recognizable detail.

FLIR K-Series Featuring the NFPA®-Compliant K65

In the heat of the battle, a thermal imaging camera is indispensable – a vital tool that helps you quickly visualize your plan of attack, locate hot spots, and save lives. Ideally, every engine and truck company should have at least one high-performance TIC on hand. Since FLIR K-Series arrived on the scene, now that’s more feasible.

Affordable K-Series TICs offer new, easier ways to see more clearly in the darkest, smokiest environments by showing big, bright FLIR images to help you maneuver more strategically, stay better oriented, and find victims faster. With greater situational awareness, you’ll improve safety and the likelihood of successful outcomes.

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FLIR KF6 Truck-Mounted Thermal Imaging Camera

KF6 Front Right

for Aerial Firefighting Applications

FLIR’s new KF6 is the industry’s first thermal imager designed and built specifically for aerial apparatus applications. Mountable at the end of a straight stick and onto a ladder bucket, the KF6 provides firefighters a high-angle thermal view of rooftops and other elevated structures, helping them see through thick smoke, more efficiently target the hottest areas, and more strategically direct firefighting efforts.

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FLIR M-Series Pan-Tilt-Zoom thermal imaging camera

MSeries Front RightPan-Tilt-Zoom TICs for ARFF, Hazmat and Rescue Trucks

FLIR’s new M-Series redefines thermal imaging for first responder vehicles, drawing on our many years of experience in building combat-proven thermal cameras for militaries and government agencies around the world.

With up to 640 × 480 thermal imaging, M-Series cameras let you see more from farther away, even in the smokiest, darkest conditions.

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FLIR Aerial First Responder Kits

Ready to Fly sUAS for Aerial Thermal Imaging

FLIR’s new Aerial First Responder Kits give you an on-demand airborne capability you’ve never had before. See through smoke to keep track of fire fighters, respond to SAR calls and other emergencies with everything you need to get in the air fast.

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The test was started on 2019-01-01

The test has been running for 584 days.

A robots noindex directive was added to the page after 40 days.

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Happy Birthday Peter Cullen - The Voice of a Cartoon Generation

Join us in wishing Peter Cullen a Happy Birthday! Born in Montreal, Canada on July 28, 1941 (age 79 years), he is the voice of a cartoon generation. Very few actors have created so many characters that touched so many lives. We all know him as the voice of Optimus Prime, but he's also stood in for many other notables suck as K.A.R.R., Eyore, Mario, Voltron, and so many others. Illustration by raggyrabbit94

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About Toa Payoh Central YEC

under construction

The Escapade is organised and managed by a team of passionate young adults from the Toa Payoh Central Community Centre Youth Executive Committee (TPY CCC YEC). Our mission is to bring the community together through fun and games.

Besides The Escapade, we are involved in a variety of volunteer projects and interest groups. If you would like to participate in any of these events, drop us a word!

We are also on a constant lookout for youth leaders to help champion initiatives that give back to our dear Singapore. If you have the heart to serve, get in touch with us!  We can use all the help we can get!

Visit our facebook page!


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1 Minute Protein Birthday Cake

by Leiann Lynn Rose Spontaneo (follow)
Self-Taught Cook. Fitness Enthusiast. Frugal Gal. Loves Her Family.
Cake (3)      1 Minute Birthday Cake (1)      Birthday Cake (1)      Birthday (1)
Oh my gosh, you forgot! What to do? You know what I am talking about... a loved one's, a friend's, a co-worker's, Birthday!

Just click on the highlighted link to learn:

How To Make A Protein Cheesecake Birthday Cake In One (1 ) Minute!

happy birthday

I am not going to repeat everything that he demonstrated, but this recipe seems too good to be true!

The cake is healthy and contains cream cheese! Usually you think Birthday cakes are so sugar-laden and fattening. Well, Birthday Girl or Birthday Boy, no sugar crashes after consumption either. So, celebrate!

Was your Birthday forgotten? Now you know how to celebrate alone.

No hot baking as obviously a no bake recipe. Totally simple!

I hope you like! Leave a comment with your thoughts!

#1 Minute Birthday Cake
#Birthday Cake
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