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USB Or Firewire links

4)A DAW software in a computer processes the digital input, owns virtual instruments playable via MIDI input, elaborates multi-track composition. Any form of digital audio can be sent back with imperceptible latency to the Soundcard to be monitored 3) The Soundcard, as a ADC, turns the analog input into a digital sound (sampling)

MIDI input via USB et similar links

Analog wires such as RCA, XLR or jack connectio n

5) The soundcard, as a DAC, turns back the digital feedback input into an analog signal 6)The mixer, once received a signal from the soundcard, send it to an amplifier (or to active speakers)

2)A mixer collects the analog sound applying a so called preamplification and sends it to a Soundcard (Amplification is not required for a line standard input such as signals from CD players) 1) A microphone catches sound waves in the air and turns them into a similar electromagneti c wave: it's the analog sound

Analog STEREO wire

7) As a mic counterpart, speakers turn an amplified signal (active speakers have built-in amplifiers) into sound Represented model: Northbrook college waves. studio Mixer: Soundcraft CompACT 40 Soundcard: digidesign Mbox2 Computer: iMac 2011 27-inch DAWs: ProTools; Logic Studio; Ableton Live MIDI Controller/s: M-Audio Oxygen 49 (Keyboard) Amplifier: Samson Servo-170 Loudspeakers: A pair of JBL Control 28

The Diagram shows the journey of a sound into a recording process. As a first step there is an analog audio signal. It can come from a phono source such as a microphone, an analog player (i.e. turntables, cassette players...) or an analog instrument (bass guitar, electric guitar, electric piano...) or from a line source (such as CD players, amplifiers, etc). If it's the first case the signal is very quiet and an amplification is needed: it must be sent to a pre-amplifier (usually built-in component of the phono-ports in mixers and in some soundcard). In our example the sound source is a voice (or an acoustic instrument, or a whatever real sound going around the air) so that the first conversion is made by a microphone due to turn air vibrations into an electric analog sound. Basically a diaphragm (or a ribbon) vibrates when sound waves come in and electrical current is produced as an imitation of the waveform. There are different kinds of mics to achieve different purposes. Very sensitive, loud-output, wide-frequency responding, great-transient responding Condenser microphones for instance. They usually require 48 phantom power, so that you have to be sure your mixer port can provide it. Condensers can be Large Diaphragm or Small Diaphragm mics. The first ones are the best choice for studio recording of vocals and any instrument you want to sound deep and warm. The second ones are generally the choice where you want a solid, wide frequency response and the best transient response. Dynamics microphones are based on a diaphragm system too. To be correct, a lightweight diaphragm, usually made of plastic film, is attached to a very small coil of wire suspended in the field of a permanent magnet. When a sound causes the diaphragm to vibrate, the whole assembly works as a miniature electricity generator, and a minute electric current is produced. In live sound, nearly all the mics used are dynamics, and in the studio, instruments such as drums, electric guitars, and basses may also be recorded using dynamic mics. Dynamic microphones have the advantages of being relatively inexpensive and hard-wearing, and they don't need a power supply or batteries to make them operate. The weakness of the dynamic mic lies in the fact that the sound energy has to move both the mic diaphragm and the wire coil attached to it. The mass of the coil adds to the inertia of the diaphragm assembly, which in turn restricts the frequency response of the microphone. In practical terms, the outcome is that dynamic microphones fail to reproduce very high frequencies accurately. In some applications, this isn't too serious, but if you're working with an instrument where a lot of tonal detail is contained in the upper harmonics, a dynamic mic is unlikely to bring out the best in that instrument. Another side-effect of the finite mass of the diaphragm/coil assembly is that the dynamic microphone is not particularly efficient -- a lot of amplification has to be used to make the signal usefully large, and the more gain you use, the more noise you add to the signal. In the studio where the mic is used very close to the sound source, this lack of efficiency is not a major problem, but if you're trying to capture a quiet or very distant sound, then a dynamic mic isn't likely to produce good results. To summarise; dynamic microphones are most effective when working with relatively loud sound sources that don't contain a lot of very high-frequency detail. They're also tough as old boots, which makes them good for live work.

Another type of dynamic microphone is the ribbon microphone, but these are only used rarely in recording applications by engineers who appreciate the subtleties of the ribbon sound. These mics are comprised of a thin metal ribbon suspended in a magnetic field, and when sound energy is encountered, the electrical signal generated is induced in the ribbon itself rather than in a voice coil. The main advantage of ribbon microphones is their smooth, detailed sound; the disadvantages are their higher cost and the fact that they are more fragile than conventional dynamic mics. A common problem with all these mics is the electrical output. Because the electrical output is so very small, it has to be amplified using a mic preamp before it is large enough to be useful. In our studio scheme, the so-called phono-to-line turning is operated by the mixer between the mic and the soundcard. 3-pin XLR connectors (but also phono jack connectors) are used as they deal with balanced audio signals. Using a balanced signal reduces the risk of inference. Through the mixer, we can also apply an equalisation to the signal to adjust the sound colour and to stress a certain frequency range. You can also use the mixer to plug other phono (but also line) sources in. Now that the sound is a line standard one, it is delivered to the soundcard through a pair (L/R) of audio wires with jack, RCA or XLR connectors (in our example we have XLR connectors) Inside the soundcard the sampling occurs. This is a core part in every digital studio. In fact, you would be able to run a small studio using only a mic, a soundcard (there are some soundcards with built-in preamp available) and your computer. Although there is a built-in soundcard in every computer, to achieve a decent quality when sampling any sound, buying a professional soundcard is inevitable. Almost every pro soundcard is an external soundcard today (audio interface), having the form of an external FireWire or USB unit, usually for convenience and improved fidelity.. It should be able to sample with a bit depth of 24-bit and a bit-rate of at least 44.1 kHz, with low-latency audio recording-playback. A good soundcard must deal with multichannel recording-playback in realtime and one of the limitations of consumer soundcards is their comparatively large sampling latency; this is the time it takes for the AD Converter to complete conversion of a sound sample and transfer it to the computer's main memory. Consumer soundcards are also limited in the effective sampling rates and bit depths they can actually manage and have lower numbers of less flexible input channels: professional studio recording use typically requires more than the two channels that consumer soundcards provide, and more accessible connectors, unlike the variable mixture of internaland sometimes virtualand external connectors found in consumer-grade soundcards. Once you avoided consumer products, the main question you have to ask yourself is: how many channels do I need?. Immediately after comes do I need built-in preamps or phantom power?. Prices and quality are more or less similar among soundcards with the same number of I/O channels and wether they have or not phantom power and/or preamps. Since, after the conversion, the sound is no longer a physical wave, but a digital information code, it must be sent to a computer, via digital data wires. The connection must be fast as the soundcard does not create the digital audio file nor store anything

while sampling, so that the cpu is quite involved into the process. Cards meeting the USB 1.1 specification have sufficient data transfer capacity to support high quality sound operation if their circuit design permits. However, USB 1.1 data throughput allows for concurrent use of a limited number of channels and/or limited sampling frequency and/or limited bit depth. For example 2 channels in and 2 channels out at 48 kHz/16 bit, or 2 channels out and no input at 96 kHz/24-bit. For a higher number of channels and/or digital signal parameters, USB 2.0 or higher or Firewire standards are needed. Once sampled, how the sound get recorded? A software must deal with it. Because the quality of the sound depends on the way it happens, for a professional recording a good software is needed. There are plenty of programs which can record a digital file to be available in a series of formats as it get stored in the pc hard drive; some of them are even freeware and this doesn't affect the quality at all. Anyway a studio engineer who wants not only to record a file, but also to edit and use it to create a more complex music tune must look at a more advanced solution: this solution is called Digital Audio Workstation, also known as DAW. DAWs today are incredibly developed, allowing you to process audio data at high quality, mixing several tracks like an analog studios if not better. A DAW is the true heart of a studio nowadays, from the biggest ones to the smallest, at the point that you could even create a song just using a computer with a good DAW. In fact a Daw, rather than to deal only with the editing and the mixing of audio tracks, uses the MIDI code to play virtual instrument (which sounds have been sampled before), virtual synthesizers (which literarily creates, through a series of algorithms, sound waves), to customise them and to store presets of the patches you create to achieve a certain sound. Not only that, a DAW allows you to master your tune adding effects to control the dynamic aspect, the filtering, the echoes, and so on, permitting the virtual recreation of natural sound fields but also to experiment with digital effects. Creating a tune using a software like a DAW is a process called Sequencing. Best known DAWs include Live (one of the most useful if you have to work with clip sequencing), Logic (the Apple brand sequencer), Sonar (the PC dedicated sequencing solution), Pro Tools and Cubase (most used for recording and mixing audio files), and many others. DAWs have also the advantage of supporting MIDI code, so that you can control their features and parameters via external controllers. In our example a MIDI keyboard have been plugged into the computer via USB so that you can play the sequencer easily and in real time. Alternatively, a DAW can read a MIDI signal sent by a MIDI interface where you can plug in any kind of midi source through MIDI wires. With a good processor and a good Soundcard you can even monitor your project in real time using earphones, loudspeakers or both. The processed sound in fact is sent back, with an imperceptible latency, to the soundcard through the data wire which connects the computer to the soundcard. The soundcard then converts back the digital signal to an analog one and sends it to the monitors. In our example the soundcard sends the stereo sound to an Amplifier linked to a pair of Loudspeakers which receive respectively the left audio signal and the right audio signal to reproduce it. Alternative you can plug earphones into the soundcard to monitor the sound

or you can send the unamplified sound directly to a pair of loudspeakers if they are actives ones (with a built in amplifier). The best tip to monitor the sound is basically not to equalise the signal, while using a neutral reproduction sound system, to obtain a good fidelity to the final mix. As a matter of fact such a studio can record audio sources while monitoring it (turning anoalog audio into digital audio), add virtual sounds (like digital instruments you can play both writing MIDI notes or playing MIDI controllers), mix and process the ensemble (controlling each track separately) and listen to it (turning back the digital audio into analog audio) in real time, while saving top quality clips or tunes as digital files! That's why digital recording is used in every studio solution today.