Saturday, June 30, 2007

Live Sound Mixing




Live sound mixing is the art of combining and processing a number of audio signals together to create a "mix" that the audience or performers at a live show hear. There are two types of live sound mixing: Front of House (FOH) and Monitor mixing.

Whenever sound reinforcement is needed for a live performance of either music, theater, or spoken word, a sound system is set up to provide this reinforcement. This sound system generally comprises a number of microphones on the stage, a mixing board, a number of speakers, often a number of audio processing devices, and the cabling to connect all of these components. For smaller venues and sound systems, the performer(s) often do not need a live sound engineer to operate the system during their performance. But when the venue and complexity of the sound system reaches a certain size, at least one live sound engineer is needed to operate the system. A live sound engineer refers to a person that is experienced in the set up and operation of a sound reinforcement system.

For mid sized venues and sound systems, usually only one live sound engineer is needed to mix the sound. When only one engineer is present, both the Front of House mix and the Monitor mix are done by the one engineer with one mixing board at the Front of House position. For larger sound systems and venues, at least two engineers and a number of technicians are required to run the system. The two primary engineers are the Front of House engineer and the Monitor Engineer. The Front of House engineer mixes the sound that the audience hears in the house and the Monitor engineer mixes the sound that the performers hear on stage.

Contents [hide]
1 The monitor engineer
2 The front of house engineer
3 Set up, tear down, and techs
4 See also
5 External links



[edit] The monitor engineer

A monitor engineer and console at an outdoor eventThe Monitor engineer's role is most essential at music events as opposed to spoken word events. In most cases, each performer on stage has their own individual mix that is custom tailored by the monitor engineer to suit their audio needs. The monitor engineer is then faced with the challenge of pleasing anywhere from 4, 10, or maybe even more musicians with a good mix. Though monitor speakers are still in use today, the newest and highest quality monitor system is what is known as an In Ear Monitor (IEM) system. In Ear Monitors are those hearing aid type devices that you see your favorite rock stars wearing at their performances. These are basically a pair of headphones that are custom molded for the musicians individuals ears and therefore greatly reduces the outside noise that the performer hears. This isolation protects the musicians ears from getting damaged from the long durations of high volumes that they are subjected to on a large stage. It also allows them to hear their individual mix with more clarity. At the largest and highest budgeted of concert events, each musician is hearing their own individual in ear mix. This involves much more than simply mixing the sound, but requires a great deal of additional audio processing to increase the quality of the performer's mix and therefore encourage them to perform at their best.


[edit] The front of house engineer

An FOH console at an outdoor eventWhilst all the chaos is going on backstage with the monitor engineer, the Front of House engineer mixes the house sound for the audience at the back of the venue known as the "Front of House Position." The Front of House engineer (commonly known as the "noise boy" or "sound guy") uses a variety of processors and effects to tailor a musical and high quality mix of the performance that is being done on stage. Just as the monitor engineer is, they are constantly adjusting the volume of each instrument or voice on stage and are constantly adding and adjusting various effects for the musical requirements of the song. The efforts of the Front of House engineer often go unnoticed when the sound is good due to the fact that a well mixed show will sound natural to the general public and will not sound like anything is being done other than simple amplification of the music on stage, which is not at all the only thing going on.


[edit] Set up, tear down, and techs
The other duty that the live sound engineer serves is the set up and tear down of these sound reinforcement systems. For large tours and events, this is a long (sometimes multiple day) and strenuous process. It usually involves getting to the venue much much earlier than the performers with a semi truck load of gear and unloading and setting up this often heavy equipment quickly. Of course, the two engineers could never do this alone, as they are assisted by a number of audio techs that are responsible for maintaining the system during the show whilst the engineers focus on mixing the show and getting the best sounding mix that they can. After the show is done, the live sound engineers and techs must tear down and put away this large sound system and the reload it into the truck for the next show on the tour. The tear down process always ends up taking much less time than the set up process and usually only takes a few hours. Of course, as all this is going on with the sound system, there are many other aspects of the show going on to consider such as the concert lighting, backline, catering, artist management, merchandising, security, and audience direction just to list a few.

Live sound mixing is an artform in its own as there are a number of different ways that the mix can be done and a number of different ways that it can sound. The live sound engineer usually has a music sense of some sort so that they can make the proper decisions on how to mix different types of music and different types of songs at a concert. It is truly a field that is often overlooked by the general public yet without it, concerts would never be able to approach the size that they have reached today.

Music Sequencer




In the field of electronic music, a sequencer is a device or piece of software that allows the user to record, play back and edit MIDI data. Sequencers do not record audio, only the control information for synthesizers to recreate the composition. Though the term 'sequencer' is today used primarily for software, some hardware synthesizers and almost all music workstations include a built-in MIDI sequencer, while drum machines generally have a step sequencer built in. There are still also standalone hardware MIDI sequencers, though the market demand for those has diminished greatly in the last ten years.

Many sequencers have features for limited music notation, and most are able to show music in a piano roll notation. (For software designed specifically for music notation, see scorewriter.)

Music can also be sequenced using trackers such as ModPlug Tracker, and some of those are able to sequence MIDI events too.

Contents [hide]
1 History
2 Step sequencers
3 List of software sequencers / DAWs with sequencing features
4 Hardware music sequencers
5 External links



[edit] History
Early analog music sequencers used a control voltage/trigger interface, but were replaced by digital hardware- or software-based MIDI sequencers, which play back MIDI events and MIDI control information at a specified number of beats per minute.

As computer speeds increased in the 1990s, audio recording, audio editing, and sample triggering features were added to the software. Software so enhanced is called a digital audio workstation (DAW) to distinguish from sequencers and multitrack recording programs. DAWs almost always include sequencing features but, strictly speaking, go beyond what a sequencer is.


[edit] Step sequencers
A special case of sequencers are step sequencers. Instead of recording played notes or drawing notes by hand on the piano roll, the user composes patterns using a grid of (usually) 16 buttons, or steps, each step being 1/16th of a measure. Step sequencer patterns are monophonic by nature, but usually a single pattern may contain individual subpatterns for a number of different instruments. These patterns are then chained together to form longer compositions. Step sequencers are mostly used in drum machines and grooveboxes.


[edit] List of software sequencers / DAWs with sequencing features

Cakewalk's Home Studio 2002.
Steinberg's Cubase VST.For a list of trackers, see the tracker article.
Also see: List of MIDI editors and sequencers.

Commercial Sequencers:

Acid and Cinescore from Sony
Cubase and Nuendo from Steinberg
Digital Performer and AudioDesk from MOTU
FL Studio from Image Line Software
Live from Ableton
Logic Pro, Logic Express and Garage Band from Apple
Pro Tools from Digidesign
Reason from Propellerhead
Samplitude, Sequoia, Music Maker and Music Studio from Magix
SAWStudio from RML Labs [1]
Sonar, Project5 and Home Studio from Cakewalk
Storm from Arturia
Tracktion from Mackie
Open Source Sequencers:

Jazz++ [2]
LMMS [3]
MusE
Rosegarden
Seq24 [4]
Hydrogen

[edit] Hardware music sequencers
In alphabetical order (and by no means exhaustive):

AKAI MPC series
Alesis MMT-8
Clavivox, keyboard synth patented in 1956 by Raymond Scott
Doepfer MAQ 16-3
Doepfer Schaltwerk
Doepfer Regelwerk
Ensoniq ASR-10
Ensoniq ESQ-1
Ensoniq EPS-16
Fairlight CMI
Frostwave Fat Controller
genoQs Octopus
Infection Music Phaedra
Infection Music Zeit
Latronic Notron
Kawai Q-80
Korg SQ-8
Korg SQD-1
Korg SQD-8
Manikin Schrittmacher
Moog 960 Sequential Controller -- part of the Moog modular synthesizer system, and possibly the earliest sequencer.
Radikal Technologies Spectralis
RCA Mark II Sound Synthesizer (Victor). Room-filling device built in 1957 for a half-million dollars. Included a 4-polyphony synth with 12 oscillators, a sequencer fed with paper tape, and a shellac record lathe for output.
Roland MC-4
Roland MC-8
Roland MC-300
Roland MC-303
Roland MC-327
Roland MC-50
Roland MC-50 Mk2
Roland MC-500 Microcomposer
Roland MC-505
Roland MC-808
Roland MC-909
Roland MV-30
Roland MV-8000
Roland SB-55
Roland TB-303
Sequential Circuits PolySequencer
Sequentix P3
Yamaha PSR-3000
Yamaha QX1
Yamaha QX3
Yamaha QX5
Yamaha QX7
Yamaha QX21
Yamaha QY10
Yamaha QY300
Yamaha QY700
Yamaha QY100
Yamaha RM1x
Yamaha RS7000
Zyklus MPS

Musical Instrument Digital Interface (MIDI)




MIDI (Musical Instrument Digital Interface; IPA: /ˈmɪdi/) is an industry-standard electronic communications protocol that enables electronic musical instruments, computers and other equipment to communicate, control and synchronize with each other in real time.


Note names and MIDI note numbers.MIDI does not transmit an audio signal or media — it simply transmits digital data "event messages" such as the pitch and intensity of musical notes to play, control signals for parameters such as volume, vibrato and panning, cues and clock signals to set the tempo. As an electronic protocol, it is notable for its success, both in its widespread adoption throughout the industry, and in remaining essentially unchanged in the face of technological developments since its introduction in 1983. Also see: Category:MIDI standards

Contents [hide]
1 History
2 Overview
3 MIDI interfaces
4 MIDI message interoperability
5 How MIDI channel messages work
6 How MIDI Show Control works
7 The MIDI 1.0 Protocol
7.1 Hardware Transport (Electrical and Mechanical Connections)
7.2 Message Format
7.2.1 Low bandwidth
8 MIDI file formats
8.1 Standard MIDI File (SMF) Format
8.2 MIDI Karaoke File (.KAR) Format
8.3 XMF File Formats
8.4 RMI File Format
9 MIDI usage and applications
9.1 Extensions of the MIDI standard
9.1.1 General MIDI
9.1.2 General MIDI 2
9.1.3 SP-MIDI
9.1.4 Alternate Hardware Transports
9.1.5 Alternate Tunings
9.2 Other applications of MIDI
9.3 MIDI controllers: hardware, software, datastream
10 Beyond MIDI 1.0
10.1 OSC
10.2 mLAN
10.3 HD-MIDI
11 MIDI software
12 Sound samples
13 See also
14 External links
14.1 Official MIDI Standards Organizations
14.2 Unofficial Sources
14.3 MIDI Search engines
14.4 Other resources



[edit] History
By the end of the 1970s, electronic musical devices were becoming increasingly common and affordable. However, devices from different manufacturers were generally not compatible with each other and could not be interconnected. Different interfacing models included:

analog control voltages at various standards (such as 1 volt per octave, or the logarithmic "hertz per volt")
analog clock, trigger and "gate" signals (both positive "V-trig" and negative "S-trig" varieties, between -15V to +15V)
proprietary digital interfaces such as Roland Corporation's DCB (digital control bus) and Yamaha's "keycode" system.
In an attempt to find a way forward from this situation, audio engineer and synthesizer designer Dave Smith of Sequential Circuits, Inc. proposed the MIDI standard in 1981 in a paper to the Audio Engineering Society. The proposal received widespread enthusiasm within the industry, and the MIDI Specification 1.0 was published in August 1983. Today, Dave Smith is generally regarded as the "Father of MIDI" and MIDI technology has been standardized and is maintained by the MIDI Manufacturers Association (MMA).


[edit] Overview
All official MIDI standards are jointly developed and published by the MIDI Manufacturers Association (MMA) in Los Angeles, California, USA (http://www.midi.org), and for Japan, the MIDI Committee of the Association of Musical Electronic Industry (AMEI) in Tokyo (http://www.amei.or.jp). The primary reference for MIDI is The Complete MIDI 1.0 Detailed Specification, document version 96.1, available only directly from MMA in English, or from AMEI in Japanese.

The MIDI Show Control (MSC) protocol (in the Real Time System Exclusive subset) is an industry standard ratified by the MIDI Manufacturers Association in 1991 which allows all types of media control devices to talk with each other and with computers to perform show control functions in live and canned entertainment applications. Just like musical MIDI (above), MSC does not transmit the actual show media — it simply transmits digital data providing information such as the type, timing and numbering of technical cues called during a multimedia or live theatre performance.

Almost all music recordings today use MIDI devices. In addition, MIDI is also used to control hardware including recording devices and live performance equipment such as stage lights and effects pedals.

MIDI allows computers, synthesizers, MIDI controllers, sound cards, samplers and drum machines to control one another, and to exchange system data.

MIDI was a major factor in bringing an end to the "wall of synthesizers" phenomenon in 1970s-80s rock music concerts, when keyboard instrument performers were sometimes hidden behind banks of various instruments. Following the advent of MIDI, many synthesizers were released in rack-mount versions, enabling performers to control multiple instruments from a single keyboard.

Another important result of MIDI has been the development of hardware and computer-based sequencers, which can be used to record, edit and play back performances. In the years immediately after the 1983 ratification of the MIDI specification, MIDI interfaces were released for both the Apple Macintosh computer and the Windows platform, allowing for the development of a market for powerful, inexpensive, and now-widespread computer-based MIDI sequencers.

Synchronization of MIDI sequences is made possible by the use of MIDI timecode, an implementation of the SMPTE time code standard using MIDI messages, and MIDI timecode has become the standard for digital music synchronization.

A number of music file formats have been based on the MIDI bytestream. These formats are very compact; a file as small as 10 KB can produce a full minute of music. This is advantageous for applications such as mobile phone ringtones, and some video games.

The term "MIDI sound" has often been used as a synonym for "bad sounding computer music", but this reflects a misunderstanding: MIDI does not define the sound, only the control protocol. This is probably a result of the poor quality sound sythesis provided by many early sound cards, which relied on FM synthesis instead of wavetables to produce audio.


[edit] MIDI interfaces
All MIDI In and MIDI Out connectors are part of a MIDI interface. A MIDI interface moves internal binary data to the MIDI Out connector for transmission to another device's MIDI In connector, in MIDI message form. It also receives incoming MIDI messages arriving on the MIDI In connector (from another device's MIDI Out connector) into internal binary data. All MIDI compatible instruments have a built-in MIDI interface. Some computers' sound cards have a built-in MIDI Interface, whereas others require an external MIDI Interface which is usually connected to the computer via USB or FireWire.


[edit] MIDI message interoperability
All MIDI compatible controllers, musical instruments, and MIDI-compatible software follow the same MIDI 1.0 specification, and thus interpret any given MIDI message the same way, and so can communicate with and understand each other. For example, if a note is played on a MIDI controller, it will sound at the right pitch on any MIDI instrument whose MIDI In connector is connected to the controller's MIDI Out connector. Often, the joystick port doubles as a midi port.


[edit] How MIDI channel messages work
When a musical performance is played on an MIDI instrument (or controller) it transmits MIDI channel messages from its MIDI Out connector. A typical MIDI channel message sequence corresponding to a key being struck and released on a keyboard is:

The user presses the middle C key with a specific velocity (which is usually translated into the volume of the note but can also be used by the synthesiser to set the timbre as well). ---> The instrument sends one Note On message.
The user changes the pressure applied on the key while holding it down - a technique called aftertouch (can be repeated, optional). ---> The instrument sends one or more Aftertouch messages.
The user releases the middle C key, again with the possibility of velocity of release controlling some parameters. ---> The instrument sends one Note Off message.
Note On, Aftertouch, and Note Off are all channel messages. For the Note On and Note Off messages, the MIDI specification defines a number (from 0–127) for every possible note pitch (C, C#, D etc.), and this number is included in the message. For example, the Middle C note played on any MIDI compatible musical instrument will always transmit the same MIDI channel message from its MIDI Out connector.

Other performance parameters can be transmitted with channel messages, too. For example, if the user turns the pitch wheel on the instrument, that gesture is transmitted over MIDI using a series of Pitch Bend messages (also a channel message). The musical instrument generates the messages autonomously; all the musician has to do is play the notes (or make some other gesture that produces MIDI messages). This consistent, automated abstraction of the musical gesture could be considered the core of the MIDI standard.


[edit] How MIDI Show Control works
Main article: MIDI Show Control.

When any cue is called by a user (typically a Stage Manager) and/or preprogrammed timeline in a show control software application, the show controller transmits one or more Real Time System Exclusive messages from its 'MIDI Out' port. A typical MSC message sequence is:

the user just called a cue
the cue is for lighting device 3
the cue is number 45.8
the cue is in cue list 7

[edit] The MIDI 1.0 Protocol
Main article: The MIDI 1.0 Protocol
IMPORTANT: Some of the information in this section diverges from the official MMA/AMEI MIDI specifications in terminology and in technical detail. Developers interested in maximizing interoperability are encouraged to work directly from the official MMA/AMEI specifications.

There are two sides to MIDI 1.0: the hardware transport specification describing the electrical and mechanical connection, and the message format specification.


[edit] Hardware Transport (Electrical and Mechanical Connections)

MIDI ports and cable.The MIDI standard consists of a communications messaging protocol designed for use with musical instruments, as well as a physical interface standard. It consists physically of a one-way (simplex) digital current loop serial communications electrical connection signaling at 31,250 bits per second. One start bit (must be 0), eight data bits, no parity bit and one stop bit (must be 1) is used.


[edit] Message Format
Every MIDI connection is a one-way connection from the MIDI Out connector of the sending device to the MIDI In connector of the receiving device. Each such connection can carry a stream of MIDI messages, with most messages representing a common musical performance event or gesture such as note-on, note-off, controller value change (including volume, pedal, modulation signals, etc.), pitch bend, program change, aftertouch, channel pressure. All of those messages include channel number. There are 16 possible channels in the protocol. The channels are used to separate "voices" or "instruments", somewhat like tracks in a multi-track mixer.


[edit] Low bandwidth
MIDI messages are extremely compact, due to the low bandwidth of the connection, and the need for real-time accuracy. Most messages consist of a status byte (channel number in the low 4 bits, and an opcode in the high 4 bits), followed by one or two data bytes. However, the serial nature of MIDI messages means that long strings of MIDI messages take an appreciable time to send, at times even causing audible delays, especially when dealing with dense musical information or when many channels are particularly active.

To further optimize the data stream, "Running status", a convention that allows the status byte to be omitted if it would be the same as that of the previous message, helps to mitigate bandwidth issues somewhat.


[edit] MIDI file formats

[edit] Standard MIDI File (SMF) Format
MIDI messages (along with timing information) can be collected and stored in a computer file system, in what is commonly called a MIDI file, or more formally, a Standard MIDI File (SMF). The SMF specification was developed by, and is maintained by, the MIDI Manufacturers Association (MMA). MIDI files are typically created using computer-based sequencing software (or sometimes a hardware-based MIDI instrument or workstation) that organizes MIDI messages into one or more parallel "tracks" for independent recording and editing. In most but not all sequencers, each track is assigned to a specific MIDI channel and/or a specific General MIDI instrument patch. Although most current MIDI sequencer software uses proprietary "session file" formats rather than SMF, almost all sequencers provide export or "Save As..." support for the SMF format.

An SMF consists of one header chunk and one or more track chunks. There are three SMF formats; the format is encoded in the file header. Format 0 contains a single track and represents a single song performance. Format 1 may contain any number of tracks, enabling preservation of the sequencer track structure, and also represents a single song performance. Format 2 may have any number of tracks, each representing a separate song performance. Sequencers do not commonly support Format 2.

Large collections of SMFs can be found on the web, most commonly with the extension .mid. These files are most frequently authored with the assumption that they will be played on General MIDI players.


[edit] MIDI Karaoke File (.KAR) Format
MIDI-Karaoke (which uses the ".kar" file extension) files are an "unofficial" extension of MIDI files, used to add synchronized lyrics to standard MIDI files. SMF players play the music as they would a .mid file but do not display these lyrics unless they have specific support for .kar messages. These often display the lyrics synchronized with the music in "follow-the-bouncing-ball" fashion, essentially turning any PC into a karaoke machine.

MIDI-Karaoke file formats are not maintained by any standardization body.


[edit] XMF File Formats
The MMA has also defined (and AMEI has approved) a new family of file formats, XMF (eXtensible Music File), some of which package SMF chunks with instrument data in DLS format (Downloadable Sounds, also an MMA/AMEI specification), to much the same effect as the MOD file format. The XMF container is a binary format (not XML-based, although the file extensions are similar). See the main article Extensible Music Format (XMF).


[edit] RMI File Format
On Microsoft Windows, the system itself uses RIFF-based MIDI files with the .rmi extension. Note, Standard MIDI Files per se are not RIFF-compliant. An RMI file, however, is simply a Standard MIDI File wrapped in a RIFF header. If the RIFF header is thrown away, the result should be a regular Standard MIDI File.

The RMI file format is not maintained by any standardization body.


[edit] MIDI usage and applications
Main article: MIDI usage and applications

[edit] Extensions of the MIDI standard
Many extensions of the original official MIDI 1.0 spec have been standardized by MMA/AMEI. Only a few of them are described here; for more comprehensive information, see the MMA web site.


[edit] General MIDI
The General MIDI (GM) and General MIDI 2 (GM2) standards define a MIDI instrument's response to the receipt of a defined set of MIDI messages. As such, they allow a given, conformant MIDI stream to be played on any conformant instrument. Although dependent on the basic MIDI 1.0 specification, the GM and GM2 specifications are each separate from it. As such, it is not generally safe to assume that any given MIDI message stream or MIDI file is intended to drive GM-compliant or GM2-compliant MIDI instruments. General Midi 1 was introduced in 1991.


[edit] General MIDI 2
Later, companies in Japan's Association of Musical Electronics Industry (sic) (AMEI) developed General MIDI Level 2 (GM2), incorporating aspects of the Yamaha XG and Roland GS formats, extending the instrument palette, specifying more message responses in detail, and defining new messages for custom tuning scales and more. The GM2 specs are maintained and published by the MMA and AMEI.

General MIDI 2 was introduced in 1992.


[edit] SP-MIDI
Later still, GM2 became the basis of the instrument selection mechanism in Scalable Polyphony MIDI (SP-MIDI), a MIDI variant for mobile applications where different players may have different numbers of musical voices. SP-MIDI is a component of the 3GPP mobile phone terminal multimedia architecture, starting from release 5.

GM, GM2, and SP-MIDI are also the basis for selecting player-provided instruments in several of the MMA/AMEI XMF file formats (XMF Type 0, Type 1, and Mobile XMF), which allow extending the instrument palette with custom instruments in the Downloadable Sound (DLS) formats, addressing another major GM shortcoming.


[edit] Alternate Hardware Transports
In addition to the original 31.25 kBaud current-loop, 5-pin DIN transport, transmission of MIDI streams over USB, IEEE 1394 AKA FireWire, and ethernet is now common. Perhaps in the long run the IETF's RTP MIDI specification for transport of MIDI streams over ethernet and internet may completely supersede the original DIN transport, since RTP MIDI is capable of providing the high-bandwidth channel that earlier alternatives to MIDI (such as ZIPI) were intended to bring. See external links below for further information.


[edit] Alternate Tunings
By convention, instruments that receive MIDI generally use the conventional 12-pitch per octave equal temperament tuning system. Unfortunately this tuning system makes many types of music inaccessible because the music depends on a different intonation system. To address this issue in a standardized manner, in 1992 the MMA ratified the MIDI Tuning Standard, or MTS. This standard allow MIDI instruments that support MTS to be tuned in any way desired, through the use of a MIDI Non-Real Time System Exclusive message.

MTS uses three bytes, which can be thought of as a three-digit number base 128, to specify a pitch in logarithmic form. The following formula gives the byte values needed to encode a given frequency in Hertz:


For a note in A440 equal temperament, this formula delivers the standard MIDI note number. Any other frequencies fill the space evenly.

While support for MTS is not particularly widespread in commercial hardware instruments, it is nonetheless supported by some instruments and software, for example the free software programs TiMidity and Scala (program), as well as other microtuners.


[edit] Other applications of MIDI
MIDI is also used every day as a control protocol in applications other than music, including:

show control
theatre lighting
special effects
sound design
recording system synchronization
audio processor control
computer networking, as demonstrated by the early first-person shooter game MIDI Maze, 1987
animatronic figure control
Such non-musical applications of MIDI are possible because any device built with a standard MIDI Out connector should in theory be able to control any other device with a MIDI In port, just as long as the developers of both devices have the same understanding about the semantic meaning of all the MIDI messages the sending device emits. This agreement can come either because both follow the published MIDI specifications, or else in the case of any non-standard functionality, because the message meanings are agreed upon by the two manufacturers.


[edit] MIDI controllers: hardware, software, datastream
The term MIDI controller is used in two different ways.

In one sense, a MIDI controller is a hardware or software entity able to transmit MIDI messages via a MIDI Out connector to other devices with MIDI In connectors.
In the other (more technical) sense, a MIDI controller is any parameter in a device with a MIDI In connector that can be set with the MIDI Control Change message. For example, a synthesizer may use controller number 18 for a low-pass filter's frequency; to open and close that filter with a physical slider, a user would assign the slider to transmit controller number 18. Then, all changes in the slider position will be transmitted as MIDI Control Change messages with the controller number field set to 18; when the synthesizer receives the messages, the filter frequency will change accordingly.

[edit] Beyond MIDI 1.0
Although traditional MIDI connections work well for most purposes, a number of newer message protocols and hardware transports have been proposed over the years to try to take the idea to the next level. Some of the more notable efforts include:


[edit] OSC
The Open Sound Control (OSC) protocol was at CNMAT. OSC has been implemented in the well-known software synthesizer Reaktor and in other projects including SuperCollider, Pure Data, Isadora, Max/MSP, Csound, VVVV and ChucK. The Lemur Input Device, a customizable touch panel with MIDI controller-type functions, also uses OSC. OSC differs from MIDI over traditional 5-pin DIN in that it can run at broadband speeds when sent over Ethernet connections. Unfortunately few mainstream musical applications and no standalone instruments support the protocol so far, making whole-studio interoperability problematic. OSC is not owned by any private company, however it is also not maintained by any standards organization.


[edit] mLAN
Yamaha has its mLAN[1] protocol, which is a based on the IEEE 1394 transport (also known as FireWire) and carries multiple MIDI message channels and multiple audio channels. mLAN is not maintained by a standards organization as it is a proprietary protocol. mLAN is open for licensing.

Audio Signal Processing

Audio signal processing
From Wikipedia, the free encyclopedia
Jump to: navigation, search
This article does not cite any references or sources.
Please help improve this article by adding citations to reliable sources. (help, get involved!)
Unverifiable material may be challenged and removed.
This article has been tagged since August 2006.
Audio signal processing, sometimes referred to as audio processing, is the processing of a representation of auditory signals, or sound. The representation can be digital or analog.

The focus in audio signal processing is most typically a mathematical analysis of which parts of the signal are audible. For example, a signal can be modified for different purposes such that the modification is controlled in the auditory domain.

The parts of the signal are heard and which are not, is not decided merely by physiology of the human hearing system, but very much by psychological properties. These properties are analysed within the field of psychoacoustics

Contents [hide]
1 History of audio processing
2 Analog signals
3 Digital signals
4 Application areas
4.1 Audio Broadcasting



[edit] History of audio processing
Audio processessing was necessary for early radio broadcasting -- as there were many problems with studio to transmitter links.


[edit] Analog signals
An analog representation is usually electrical; a voltage level represents the air pressure waveform of the sound.


[edit] Digital signals
A digital representation expresses the pressure wave-form as a sequence of symbols, usually binary numbers, which permits digital signal processing. It must be noted that all real world audio signals are continuous-time analog signals. Therefore, sampling and quantization must be applied to convert the continuous-time analog signal to a discrete-time digital representation. While such a conversion is lossy, most modern audio systems use this approach as the techniques of digital signal processing are much more powerful and efficient than analog domain signal processing.


[edit] Application areas
Processing methods and application areas include storage, level compression, data compression, transmission, enhancement (e.g., equalization, filtering, noise cancellation, echo or reverb removal or addition, etc.)


[edit] Audio Broadcasting
Audio broadcasting (be it for television or audio broadcasting) is perhaps the biggest market segement (and user area) for audio processing products -- globally.

Traditioanlly the most important audio processing (in audio brodcating) takes place just before the transmitter. Studio audio processing is limited in the modern era due to digital audio systems (mixers, routers) being pervasive in the studio.

In audio broadcasting, the audio processer must

prevent overmodulation, and minimize it when it occours
maximize overall loudness
compensate for non-lineral transmitters, more common with mediumwave and shortwave broadcasting

Audio Compression

Audio compression can mean two things:

Audio data compression - in which the amount of data in a recorded waveform is reduced for transmission. This is used in CD and MP3 encoding, internet radio, and the like.
Audio level compression - in which the dynamic range (difference between loud and quiet) of an audio waveform is reduced. This is used in guitar effects racks, recording studios, etc.

Digital Audio





A sound wave, in gray, represented digitally, in red (after a zero-order hold but before filtering)Digital audio uses digital signals for sound reproduction. This includes analog-to-digital conversion, digital-to-analog conversion, storage, and transmission.

Digital audio has emerged because of its usefulness in the recording, manipulation, mass-production and distribution of sound. Modern distribution of music across the internet through on-line stores depends on digital recording, and digital compression algorithms. Distribution of audio as data files rather than as physical objects has significantly reduced costs of distribution. However, it has brought about a rise in music sharing through peer to peer networks, which is illegal in many countries as copyright infringement. The Recording Industry Association of America and other organizations claim that music sharing severely harms the profitability of their business.

From the wax cylinder, to the compact cassette, analogue audio music storage and reproduction have been based on the same principles upon which human hearing are based.

In an analogue audio system, sounds begin as physical waveforms in the air, are transformed into an electrical representation of the waveform, via a transducer (for example, a microphone), and are stored or transmitted. To be re-created into sound, the process is reversed, through amplification and then conversion back into physical waveforms via a loudspeaker. Although its nature may change, its fundamental wave-like characteristics remain unchanged during its storage, transformation, duplication, amplification. All analogue audio signals are susceptible to noise and distortion, due to the inherent noise present in electronic circuits.

On the other hand, the digital audio chain begins when an analogue audio signal is converted into electrical signals — ‘on/off’ pulses — rather than electro-mechanical signals. This signal is then re-encoded (rather like a spy might use a code book), in order to combat any errors that might occur in the storage or transmission of the signal. It is this "channel coding" that is essential to the ability of the digital system to recreate the analogue signal upon replay. An example of a channel code is Eight to Fourteen Bit Modulation as used in the audio Compact Disc.

Contents [hide]
1 Overview of digital audio
2 Subjective evaluation
3 History of digital audio use in commercial recording
4 Digital audio technologies
5 Digital audio interfaces
6 References
7 See also



[edit] Overview of digital audio

Sampling and 4-bit quantization of an analogue signal (red) using Pulse Code Modulation.Digital audio is the method of representing audio in digital form.

An analog signal is converted to a digital signal at a given sampling rate and bit resolution; it may contain multiple channels (2 channels for stereo or more for surround sound). Generally speaking: the higher the sampling rate and bit resolution the more fidelity. Both systems introduce noise at the capturing stage, in analogue recording this is due to the noise floor of the circuit, and in digital recording due to quantization noise.

Quantization Noise (file info) — play in browser (beta)
An example of audio with progressively worsening quantization noise.
Problems listening to the file? See media help.

Much like an analog audio system, a digital audio system strives to reproduce the audio perfectly but neither can ultimately prevail. Analog systems have inherent capacitance and inductance which limit the bandwidth of the system and resistance limits the amplitude. Digital systems' sampling rate limits the bandwidth and bit resolution limits the dynamic range (resolution of amplitude creation). Both systems require increased cost and attention to achieve higher fidelity.

A digital audio signal starts with an analog-to-digital converter (ADC) that converts an analog signal to a digital signal. The ADC runs at a sampling rate and converts at a known bit resolution. For example, CD audio has a sampling rate of 44.1 kHz (44,100 samples per second) and 16-bit resolution for each channel (stereo). If the analog signal is not already bandlimited then an anti-aliasing filter is necessary before conversion, to prevent aliasing in the digital signal. (Aliasing occurs when frequencies above the Nyquist frequency have not been band limited, and instead appear as audible artifacts in the lower frequencies).

Some audio signals such as those created by digital synthesis originate entirely in the digital domain, in which case analog to digital conversion does not take place.

After being sampled with the ADC, the digital signal may then be altered in a process which is called digital signal processing where it may be filtered or have effects applied.

The digital audio signal may then be stored or transmitted. Digital audio storage can be on a CD, an iPod, a hard drive, USB flash drive, CompactFlash, or any other digital data storage device. Audio data compression techniques — such as MP3, Ogg Vorbis, or AAC — are commonly employed to reduce the size. Digital audio can be streamed to other devices.

The last step for digital audio is to be converted back to an analog signal with a digital-to-analog converter (DAC). Like ADCs, DACs run at a specific sampling rate and bit resolution but through the processes of oversampling, upsampling, and downsampling, this sampling rate may not be the same as the initial sampling rate.


[edit] Subjective evaluation
Fidelity evaluation is a long-standing issue with audio systems in general and introduction of lossy compression algorithms and psychoacoustic models have only increased debate.

Audio can be measured and analyzed to more exacting measures than can be done by ear, but what this technical measurement and analysis lacks is the ability to determine if it sounds "good" or "bad" to any given listener.[dubious — see talk page] Like any other human opinion, there are numerous parameters that widely vary between people that affect their subjective evaluation of what is good or bad. Such things that pertain to audio include hearing capabilities, personal preferences, location with respect to the speakers, and the room's physical properties.

This is not to say that subjective evaluation is unique to digital audio, digital audio can add to the fervor of discussion because it does introduce more things (e.g., lossy compression, psychoacoustic models) that can be debated.


[edit] History of digital audio use in commercial recording
Commercial digital recording of classical and jazz music began in the early 1970s, pioneered by Japanese companies such as Denon, the BBC, and British record label Decca (who in the mid-70s developed digital audio recorders of their own design for mastering of their albums), although experimental recordings exist from the 1960s. The first 16-bit PCM recording in the United States was made by Thomas Stockham at the Santa Fe Opera in 1976 on a Soundstream recorder. In most cases there was no mixing stage involved; a stereo digital recording was made and used unaltered as the master tape for subsequent commercial release. These unmixed digital recordings are still described as DDD since the technology involved is purely digital. (Unmixed analogue recordings are likewise usually described as ADD to denote a single generation of analogue recording.)

The first entirely digitally recorded (DDD) popular music album was Ry Cooder's Bop Till You Drop, recorded in late 1978. It was unmixed, being recorded straight to a two-track 3M digital recorder in the studio. Many other top recording artists were early adherents of digital recording. Others, such as former Beatles producer George Martin, felt that the multitrack digital recording technology of the early 1980s had not reached the sophistication of analogue systems. Martin used digital mixing, however, to reduce the distortion and noise that an analogue master tape would introduce (thus ADD). An early example of an analogue recording that was digitally mixed is Fleetwood Mac's 1979 release Tusk.


[edit] Digital audio technologies
DAB (Digital Audio Broadcasting)
Digital audio workstation
Digital audio player
Storage technologies:

Digital Audio Tape (DAT)
Compact disc (CD)
DVD DVD-A
MiniDisc
Super Audio CD
various audio file formats

[edit] Digital audio interfaces
Audio-specific interfaces include:

AC97 (Audio Codec 1997) interface between Integrated circuits on PC motherboards
ADAT interface
AES/EBU interface with XLR connectors
AES47, Professional AES3 digital audio over Asynchronous Transfer Mode networks
I²S (Inter-IC sound) interface between Integrated circuits in consumer electronics
MIDI low-bandwidth interconnect for carrying instrument data; cannot carry sound
S/PDIF, either over coaxial cable or TOSLINK
TDIF, Tascam proprietary format with D-sub cable
Naturally, any digital bus (e.g., USB, FireWire, and PCI) can carry digital audio.

Audio File Format

An audio file format is a container format for storing audio data on a computer system. There are numerous file formats for storing audio data.

The general approach towards storing digital audio is to sample the audio voltage (which on playback, would correspond to a certain position of the membrane in a speaker) of the individual chanels with a certain resolution (the number of bits per sample) in regular intervals (forming the sample rate). This data can then be stored uncompressed or compressed to reduce the file size.

Contents [hide]
1 Types of formats
1.1 Uncompressed audio format
1.2 Lossless audio formats
1.3 Free and Open File Formats
1.4 Open File Formats
1.5 Proprietary Formats



[edit] Types of formats
It is important to distinguish between a file format and a codec. A codec performs the encoding and decoding of the raw audio data while the data itself is stored in a file with a specific audio file format. Though most audio file formats support only one audio codec, a file format may support multiple codecs, as AVI does.

There are three major groups of audio file formats:

Uncompressed audio formats, such as WAV, AIFF and AU;
formats with lossless compression, such as FLAC, Monkey's Audio (filename extension APE), WavPack (filename extension WV), Shorten, TTA, Apple Lossless and lossless Windows Media Audio (WMA); and
formats with lossy compression, such as MP3, Vorbis, lossy Windows Media Audio (WMA) and AAC.

[edit] Uncompressed audio format
There is one major uncompressed audio format, PCM, which is usually stored as a .wav on Windows or as .aiff on Mac OS. WAV is a flexible file format designed to store more or less any combination of sampling rates or bitrates. This makes it an adequate file format for storing and archiving an original recording. A lossless compressed format would require more processing for the same time recorded, but would be more efficient in terms of space used. WAV, like any other uncompressed format, encodes all sounds, whether they are complex sounds or absolute silence, with the same number of bits per unit of time.

Let's take an example. A file contains a minute of a symphonic orchestra playing beautifully followed by a minute of silence. If the sound were stored in WAV, the same amount of data would be used for each half. If data were encoded with TTA, the first minute would be a bit smaller than in the WAV file, and the silent half would take almost no disc space at all. But then, recording in the TTA format would require a lot more processing than the WAV.

The WAV format is based on the RIFF file format, which is similar to the IFF format.

BWF (Broadcast Wave Format) is a standard audio format created by the European Broadcasting Union as a successor to WAV. BWF allows metadata to be stored in the file. See: European Broadcasting Union: Specification of the Broadcast Wave Format - A format for audio data files in broadcasting. EBU Technical document 3285, July 1997. This format is the primary recording format used in many professional Audio Workstations used in the Television and Film industry. Stand-alone file based multi-track recorders from Sound Devices, Zaxcom, HHB USA, Fostex, and Aaton all use BWF as their preferred file format for recording multi-track audio files with SMPTE Time Code reference. This standardized Time Stamp in the Broadcast Wave File allows for easy synchronization with a separate picture element.


[edit] Lossless audio formats
Lossless audio formats (such as TTA and FLAC) provide a compression ratio of about 2:1.


[edit] Free and Open File Formats
wav - standard audio file format used mainly in Windows PCs. Commonly used for storing uncompressed (PCM), CD-quality sound files, which means that they can be large in size - around 10MB per minute of music. It is less well known that wave files can also be encoded with a variety of codecs to reduce the file size (for example the GSM or mp3 codecs). Wav files use a RIFF structure.
ogg - a free, open source container format supporting a variety of codecs, the most popular of which is the audio codec Vorbis. Vorbis offers better compression than MP3 but is less popular.
flac - a lossless compression codec. You can think of lossless compression as like zip but for audio. If you compress a PCM file to flac and then restore it again it will be a perfect copy of the original. (All the other codecs discussed here are lossy which means a small part of the quality is lost). The cost of this losslessness is that the compression ratio is not good. Flac is recommended for archiving PCM files where quality is important (eg. broadcast or music use).
aiff - the standard audio file format used by Apple. It is like a wav file for the Mac.
raw - a raw file can contain audio in any codec but is usually used with PCM audio data. It is rarely used except for technical tests.
au - the standard audio file format used by Sun, Unix and Java. The audio in au files can be PCM or compressed with the ulaw, alaw or G729 codecs.

[edit] Open File Formats
mp3 - the MPEG Layer-3 format is the most popular format for downloading and storing music. By eliminating portions of the audio file that are essentially inaudible, mp3 files are compressed to roughly one-tenth the size of an equivalent PCM file while maintaining good audio quality. The mp3 format is recommended for music storage. It is not that good for voice storage.
gsm - designed for telphony use in Europe, gsm is a very practical format for telephone quality voice. It makes a good compromise between file size and quality. Note that wav files can also be encoded with the gsm codec.
dct - A variable codec format designed for dictation. It has dictation header information and can be encrypted (often required by medical confidentiality laws).
vox - the vox format most commonly uses the Dialogic ADPCM (Adaptive Differential Pulse Code Modulation) codec. Similar to other ADPCM formats, it compresses to 4-bits. Vox format files are similar to wave files except that the vox files contain no information about the file itself so the codec sample rate and number of channels must first be specified in order to play a vox file.
aac - the Advanced Audio Coding format is based on the MPEG2 and MPEG4 standards. aac files are usually ADTS or ADIF containers.
mp4/m4a - MPEG-4 audio most often AAC but sometimes MP2/MP3

[edit] Proprietary Formats
wma - the popular Windows Media Audio format owned by Microsoft. Designed with Digital Rights Management (DRM) abilities for copy protection.
atrac (.wav) - the older style Sony ATRAC format. It always has a .wav file extension. To open these files simply install the ATRAC3 drivers.
ra - a Real Audio format designed for streaming audio over the Internet. The .ra format allows files to be stored in a self-contained fashion on a computer, with all of the audio data contained inside the file itself.
ram - a text file that contains a link to the Internet address where the Real Audio file is stored. The .ram file contains no audio data itself.
dss - Digital Speech Standard files are an Olympus proprietary format. It is a fairly old and poor codec. Prefer gsm or mp3 where the recorder allows.
msv - a Sony proprietary format for Memory Stick compressed voice files.
dvf - a Sony proprietary format for compressed voice files; commonly used by Sony dictation recorders.
m4p - A proprietary version of AAC in MP4 with Digital Rights Management developed by Apple for use in music downloaded from their iTunes Music Store.
Retrieved from "http://en.wikipedia.org/wiki/Audio_file_format"