PROWRA's supporters have gone so far as to claim that “ending welfare as we . give an in-depth account of the social role marriage plays in the lives of a Making Ends Meet: How Single Mothers Survive Welfare and Low Wage Work. Download PDF and whether they and their families are able to make ends meet. Wages are discussed in depth later in the paper. Downloading YouTube videos: what you need to know original content creators - often ordinary people struggling to make ends meet. We won't go into depth over the ins and outs of online piracy here, suffice to say that in.
After accounting for demographic differences between restaurant workers and other workers, restaurant workers have hourly wages that are Unionization rates are extremely low in the restaurant industry, but unionized restaurant workers receive wages that are substantially higher than those of non-union restaurant workers. Occupations within the restaurant industry are highly gendered and have strong racial and ethnic concentrations.
Low Wages and Few Benefits Mean Many Restaurant Workers Can’t Make Ends Meet
Hispanics are disproportionately likely to be dishwashers, dining room attendants, or cooks, also relatively low-paid occupations. Restaurant workers are much more likely than other workers to be poor or near-poor. One in six restaurant workers, or The poverty rate for workers outside the restaurant industry is more than 10 percentage points lower, at 6.
Twice the official poverty threshold is commonly used by researchers as a measure of what it takes for a family to make ends meet. More than two in five restaurant workers, or Among workers in the restaurant industry, poverty rates are much lower for workers in a union. Restaurant workers rarely receive fringe benefits. Of unionized restaurant workers, The quality of restaurant jobs can be improved by reforming or enacting policies to give restaurant workers more bargaining power and raise their wages, such as: Passing comprehensive immigration reform with a path to citizenship for undocumented workers, which will make undocumented workers in the restaurant industry less vulnerable to exploitation.
Passing legislation requiring all employers to provide paid sick days to their employees. The effect is very slight, but listening tests have confirmed that both effects can be audible.
Illustration of distortion products resulting from intermodulation of a 30kHz and a 33kHz tone in a theoretical amplifier with a nonvarying total harmonic distortion THD of about.
Distortion products appear throughout the spectrum, including at frequencies lower than either tone. Inaudible ultrasonics contribute to intermodulation distortion in the audible range light blue area. Systems not designed to reproduce ultrasonics typically have much higher levels of distortion above 20kHz, further contributing to intermodulation.
Widening a design's frequency range to account for ultrasonics requires compromises that decrease noise and distortion performance within the audible spectrum. Either way, unneccessary reproduction of ultrasonic content diminishes performance. There are a few ways to avoid the extra distortion: A dedicated ultrasonic-only speaker, amplifier, and crossover stage to separate and independently reproduce the ultrasonics you can't hear, just so they don't mess up the sounds you can.
Amplifiers and transducers designed for wider frequency reproduction, so ultrasonics don't cause audible intermodulation.
Given equal expense and complexity, this additional frequency range must come at the cost of some performance reduction in the audible portion of the spectrum. Speakers and amplifiers carefully designed not to reproduce ultrasonics anyway.
- Finally, the good news
- First, the bad news
- Jobs in the restaurant industry
Not encoding such a wide frequency range to begin with. You can't and won't have ultrasonic intermodulation distortion in the audible band if there's no ultrasonic content. They all amount to the same thing, but only 4 makes any sense. If you hear anything, your system has a nonlinearity causing audible intermodulation of the ultrasonics.
Be careful when increasing volume; running into digital or analog clipping, even soft clipping, will suddenly cause loud intermodulation tones. In summary, it's not certain that intermodulation from ultrasonics will be audible on a given system. The added distortion could be insignificant or it could be noticable. Either way, ultrasonic content is never a benefit, and on plenty of systems it will audibly hurt fidelity. On the systems it doesn't hurt, the cost and complexity of handling ultrasonics could have been saved, or spent on improved audible range performance instead.
Sampling fallacies and misconceptions Sampling theory is often unintuitive without a signal processing background. It's not surprising most people, even brilliant PhDs in other fields, routinely misunderstand it.
It's also not surprising many people don't even realize they have it wrong. Sampled signals are often depicted as a rough stairstep red that seems a poor approximation of the original signal. However, the representation is mathematically exact and the signal recovers the exact smooth shape of the original blue when converted back to analog. The most common misconception is that sampling is fundamentally rough and lossy. A sampled signal is often depicted as a jagged, hard-cornered stair-step facsimile of the original perfectly smooth waveform.
If this is how you envision sampling working, you may believe that the faster the sampling rate and more bits per samplethe finer the stair-step and the closer the approximation will be. The digital signal would sound closer and closer to the original analog signal as sampling rate approaches infinity. Similarly, many non-DSP people would look at the following: Or, that as audio frequency increases, the sampled quality falls and frequency response falls off, or becomes sensitive to input phase.
These beliefs are incorrect! All signals with content entirely below the Nyquist frequency half the sampling rate are captured perfectly and completely by sampling; an infinite sampling rate is not required.
Sampling doesn't affect frequency response or phase. The analog signal can be reconstructed losslessly, smoothly, and with the exact timing of the original analog signal. So the math is ideal, but what of real world complications? The most notorious is the band-limiting requirement. Signals with content over the Nyquist frequency must be lowpassed before sampling to avoid aliasing distortion; this analog lowpass is the infamous antialiasing filter.
Antialiasing can't be ideal in practice, but modern techniques bring it very close. Oversampling Sampling rates over 48kHz are irrelevant to high fidelity audio data, but they are internally essential to several modern digital audio techniques.
Oversampling is the most relevant example [ 7 ]. Oversampling is simple and clever. You may recall from my A Digital Media Primer for Geeks that high sampling rates provide a great deal more space between the highest frequency audio we care about 20kHz and the Nyquist frequency half the sampling rate. This allows for simpler, smoother, more reliable analog anti-aliasing filters, and thus higher fidelity. This extra space between 20kHz and the Nyquist frequency is essentially just spectral padding for the analog filter.
That's only half the story. Because digital filters have few of the practical limitations of an analog filter, we can complete the anti-aliasing process with greater efficiency and precision digitally. The very high rate raw digital signal passes through a digital anti-aliasing filter, which has no trouble fitting a transition band into a tight space.
After this further digital anti-aliasing, the extra padding samples are simply thrown away.
24/192 Music Downloads
Oversampled playback approximately works in reverse. This means we can use low rate Nearly all of today's analog-to-digital converters ADCs and digital-to-analog converters DACs oversample at very high rates.
Few people realize this is happening because it's completely automatic and hidden. Thirty years ago, some recording consoles recorded at high sampling rates using only analog filters, and production and mastering simply used that high rate signal. The digital anti-aliasing and decimation steps resampling to a lower rate for CDs or DAT happened in the final stages of mastering. This may well be one of the early reasons 96kHz and kHz became associated with professional music production [ 8 ].
What about 16 bit vs.
It's true that 16 bit linear PCM audio does not quite cover the entire theoretical dynamic range of the human ear in ideal conditions. Also, there are and always will be reasons to use more than 16 bits in recording and production. None of that is relevant to playback; here 24 bit audio is as useless as kHz sampling. The good news is that at least 24 bit depth doesn't harm fidelity.
It just doesn't help, and also wastes space. Revisiting your ears We've discussed the frequency range of the ear, but what about the dynamic range from the softest possible sound to the loudest possible sound? One way to define absolute dynamic range would be to look again at the absolute threshold of hearing and threshold of pain curves.
The distance between the highest point on the threshold of pain curve and the lowest point on the absolute threshold of hearing curve is about decibels for a young, healthy listener. For reference purposes, a jackhammer at one meter is only about dB.
The absolute threshold of hearing increases with age and hearing loss. Interestingly, the threshold of pain decreases with age rather than increasing. The hair cells of the cochlea themselves posses only a fraction of the ear's dB range; musculature in the ear continuously adjust the amount of sound reaching the cochlea by shifting the ossicles, much as the iris regulates the amount of light entering the eye [ 9 ].
This mechanism stiffens with age, limiting the ear's dynamic range and reducing the effectiveness of its protection mechanisms [ 10 ]. Environmental noise Few people realize how quiet the absolute threshold of hearing really is. The very quietest perceptible sound is about -8dbSPL [ 11 ].
24/ Music Downloads are Very Silly Indeed
Using an A-weighted scale, the hum from a watt incandescent light bulb one meter away is about 10dBSPL, so about 18dB louder. The bulb will be much louder on a dimmer. This is the baseline for an exceptionally quiet environment, and one reason you've probably never noticed hearing a light bulb.
Many believe that 16 bit audio cannot represent arbitrary sounds quieter than dB. I have linked to two 16 bit audio files here; one contains a 1kHz tone at 0 dB where 0dB is the loudest possible tone and the other a 1kHz tone at dB. How is it possible to encode this signal, encode it with no distortion, and encode it well above the noise floor, when its peak amplitude is one third of a bit? Part of this puzzle is solved by proper dither, which renders quantization noise independent of the input signal.
By implication, this means that dithered quantization introduces no distortion, just uncorrelated noise. That in turn implies that we can encode signals of arbitrary depth, even those with peak amplitudes much smaller than one bit [ 12 ]. However, dither doesn't change the fact that once a signal sinks below the noise floor, it should effectively disappear. How is the dB tone still clearly audible above a dB noise floor? Our dB noise floor figure is effectively wrong; we're using an inappropriate definition of dynamic range.
As each hair cell hears only a fraction of the total noise floor energy, the noise floor at that hair cell will be much lower than the broadband figure of dB. Thus, 16 bit audio can go considerably deeper than 96dB. With use of shaped dither, which moves quantization noise energy into frequencies where it's harder to hear, the effective dynamic range of 16 bit audio reaches dB in practice [ 13 ], more than fifteen times deeper than the 96dB claim.
This assures that linear 16 bit PCM offers higher resolution than is actually required. It is also worth mentioning that increasing the bit depth of the audio representation from 16 to 24 bits does not increase the perceptible resolution or 'fineness' of the audio. It only increases the dynamic range, the range between the softest possible and the loudest possible sound, by lowering the noise floor.
However, a bit noise floor is already below what we can hear. When does 24 bit matter? Professionals use 24 bit samples in recording and production [ 14 ] for headroom, noise floor, and convenience reasons.
It does not span the entire possible signal range of audio equipment. The primary reason to use 24 bits when recording is to prevent mistakes; rather than being careful to center 16 bit recording-- risking clipping if you guess too high and adding noise if you guess too low-- 24 bits allows an operator to set an approximate level and not worry too much about it.
Missing the optimal gain setting by a few bits has no consequences, and effects that dynamically compress the recorded range have a deep floor to work with. An engineer also requires more than 16 bits during mixing and mastering. Modern work flows may involve literally thousands of effects and operations.From The Depths - RMS Titanic