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Even after getting the signal converted to bits, it is still difficult to record; the hardest part is finding a scheme that can record the bits fast enough to keep up with the signal. For example, to record two channels of audio at For optical disc recording technologies such as CDs or DVDs , a laser is used to burn microscopic holes into the dye layer of the medium.
A weaker laser is used to read these signals. This works because the metallic substrate of the disc is reflective, and the unburned dye prevents reflection while the holes in the dye permit it, allowing digital data to be represented.
The number of bits used to represent a sampled audio wave the word size directly affects the resulting noise in a recording after intentionally added dither , or the distortion of an undithered signal [18]. The number of possible voltage levels at the output is simply the number of levels that may be represented by the largest possible digital number the number 2 raised to the power of the number of bits in each sample.
If there are more bits in each sample the waveform is more accurately traced, because each additional bit doubles the number of possible values. The distortion is roughly the percentage that the least significant bit represents out of the average value. Distortion as a percentage in digital systems increases as signal levels decrease, which is the opposite of the behavior of analog systems. The sample rate is just as important a consideration as the word size.
If the sample rate is too low, the sampled signal cannot be reconstructed to the original sound signal. To overcome aliasing, the sound signal or other signal must be sampled at a rate at least twice that of the highest frequency component in the signal. This is known as the Nyquist-Shannon sampling theorem.
For recording music-quality audio the following PCM sampling rates are the most common: When making a recording, experienced audio recording and mastering engineers will normally do a master recording at a higher sampling rate i. In addition it is nowadays also possible and common to release a high resolution recording directly as either an uncompressed WAV or lossless compressed FLAC file [20] usually at 24 bits without down-converting it.
The information for digital audio is contained in a bunch of numbers which indicate the loudness or volume of the sound at a specific time. The sample rate tells you how many times per second the loudness value is captured. This number needs to be at least two times higher than the highest audible frequency, otherwise the computer will perceive high frequencies as being lower than they actually are.
Most audio is captured at The word length tells you how many numbers can be used to represent different volumes of loudness. CDs represent audio with a word length of 16 bits, allowing for different values for loudness. Most audio interfaces are capable of recording audio with a bit word length, allowing for exquisite detail.
There are some newer systems which allow for recording with a bit word length but these are, for the majority part, not available at low-cost to consumers. I would like to add a quick word about USB. There is a stigma, in the business , against USB audio interfaces. Many interfaces employ connectors with higher bandwidth, like FireWire and Thunderbolt, and charge a premium for it.
It may seem logical, faster connection, better quality audio. This is to say, USB can handle bit audio with a 96 kHz sample rate, no problem. If you notice latency in your system, it is from the digital-to-analog and analog-to-digital converters as well as the speed of your computer; latency in your recording setup has nothing to do with what connector your interface uses.
One last thing before we move on to the DAW, I mentioned earlier that frequencies above half the recording sample rate will be perceived, by your computer, as lower frequencies. These lower frequencies can show up in your recording and can cause distortion.
For this reason, it is often advantageous to record at higher sample rates to avoid having these higher frequencies perceived within the audible range. Most audio interfaces allow for recording bit audio with a 96 kHz sample rate. The digital audio workstation, or DAW for short, is perhaps the most flexible element of your home-studio. There are many many many DAW software packages out there, ranging in price and features. For those of you looking to just get into audio recording, Audacity is a great DAW to start with.
This software is free and simple. It offers many built-in effects and can handle the full recording capability of any audio interface which is to say, if you record something well on this simple and free software, it will sound mighty good. This is a fancy way of saying that once you make a change to your recorded audio, you might not be able to un-make it.
This allows you to play around a lot more with your sound after its recorded. More expensive DAWs also tend to come with a better-sounding set of built-in effects. This is most noticeable with more subtle effects like reverb. My recommendation is to shop around until something catches your eye. The microphone, for many people, is the most fun part of recording! They come in many shapes and sizes and color your sound more than any other component in your setup.
Two different microphones can occupy polar opposites in the sonic spectrum. There are two common types of microphones out there: I can get carried away with physics sometimes so I will try not to write too much about this particular topic. Condenser microphones are a more recent invention and offer the best sound quality of any microphone.
They employ a charged parallel plate capacitor to measure vibrations in the air. Because of the nature of their design, condenser microphones require a small amplifier circuit built-into the microphone. Most new condenser microphones use a transistor-based circuit in their internal amplifier but older condenser mics employed internal vacuum-tube amplifiers; these tube microphones are among some of the clearest and most detailed sounding microphones ever made.
Dynamic microphones, like condenser microphones, also come in two varieties, both emerging from different eras. The ribbon microphone is the earlier of the two and observes sound with a thin metal ribbon suspended in a magnetic field. These ribbon microphones are fragile but offer a warm yet detailed quality-of-sound. The more common vibrating-coil dynamic microphone is the most durable and is used most often for live performance. The prevalence of the vibrating-coil microphone means that the vibrating-coil is often dropped from the name sometimes the dynamic is also dropped from the name too ; when you use the term dynamic mic, most people will assume you are referring to the vibrating-coil microphone.
With the wonders of globalization, all microphones can be purchase at similar costs. This means you can use many brushes to paint your sonic picture. The pits disrupt this reflection and yield up the data. In either case, the process is helped by avoiding numbers that are hard to detect, like That example is difficult because it will give just a single very short electrical spike. If some numbers are unusable, a larger maximum more bits must be available to allow recording the entire set.
On tape, twenty bits are used to record each sixteen bit sample, on CDs, twenty-eight bits are used.
Even with these techniques, the bits are going to be physically very small, and it must be assumed that some will be lost in the process. A single bit can be very important suppose it represents the sign of a large number! Error correction is really two problems; how to detect an error, and what to do about it. The most common error detection method is parity computation. An extra bit is added to each number which indicates whether the number is even or odd.
When the data is read off the tape, if the parity bit is inappropriate, something has gone wrong. This works well enough for telephone conversations and the like, but does not detect serious errors very well. In digital recording, large chunks of data are often wiped out by a tape dropout or a scratch on the disk.
Catching these problems with parity would be a matter of luck. To help deal with large scale data loss, some mathematical computation is run on the numbers, and the result is merged with the data from time to time.
Then I set the sampling rate for if I am doing a voice file. Improving performance is usually only a matter of increasing the word size or the sample rate, which is achieved by duplicating elements of the circuit. Get fast, free shipping with Amazon Prime. To read the data, light from a gentler laser is reflected off the surface of the plastic from the back: This is best format most systems can record to.
If a mistake turns up in this number, an error has occurred since the last correct CRCC was received. Once an error is detected, the system must deal gracefully with the problem. To make this possible, the data is recorded in a complex order. Instead of word two following word one, as you might expect, the data is interleaved, following a pattern like:. With this scheme, you could lose eight words, but they would represent several isolated parts of the data stream, rather than a large continuous chunk of waveform. When a CRC indicates a problem, the signal can be fixed. For minor errors, the CRCC can be used to replace the missing numbers exactly.
If the problem is more extensive, the system can use the previous and following words to reconstruct a passable imitation of the missing one. One of the factors that makes up the price difference in various digital systems is the sophistication available to reconstruct missing data.
Find the whole idea of digital audio to be a bit confusing? In this beginner's post, I explain it so it's simple enough for anyone to understand. Basics of Digital Recording. CONVERTING SOUND INTO NUMBERS. In a digital recording system, sound is stored and manipulated as a stream of discrete.
You may be wondering about the point of all of this, if it turns out that a digital system is more complex than the equivalent analog circuit. Digital circuits are complex, but very few of the components must be precise; most of the circuitry merely responds to the presence or absence of current.
Improving performance is usually only a matter of increasing the word size or the sample rate, which is achieved by duplicating elements of the circuit. It is possible to build analog circuits that match digital performance levels, but they are very expensive and require constant maintenance. The bottom line is that good digital systems are cheaper than good analog systems.