P2 Music Exercise

P2 Understand the methods and principles of sound design and production

Research and explain when you would use the following sound format and why?

Sound file formats:

Uncompressed: e.g. WAV, AIFF, AU, SMP, VOC, PCM

Lossy compression: e.g. MP3, RA, and VOX, VORBIS.
XMA

Lossless Compression:
Lossless compression data reduction methods also reduce file size, but retain all of the original data or music. Lossless compression is used where the quality of the compressed music file is critical. Another example of lossless compression is data or text where all of the information must be retained for accuracy. DTS-HD and Dolby True HD are two examples of lossless compression. Lossless compression or Uncompressed file types are used when the developer wants the highest quality of sound. .WAV files are often used for short sound effects in video games such as a character grunting when hurt.

WAV: Waveform Audio File Format (WAVE, or more commonly known as WAV due to its filename extension, is a Microsoft and IBM audio file format standard for storing an audio bit stream on PCs. It is so wide spread today that it is called a standard PC audio file format. Used primarily in PCs, the Wave file format has been accepted as a viable interchange medium for other computer platforms, such as Macintosh. This allows content developers to freely move audio files between platforms for processing. This is an example of what a WAV format could contain; The Wave file format stores information about the file's number of tracks (mono or stereo), sample rate, bit depth, as well as the uncompressed raw audio data.

• Can be played by nearly all Windows applications that support sound
• Fast decoding
• Very large file size

AIFF: The format was developed by Apple Inc. in 1988 based on Electronic Arts' Interchange File Format (IFF, widely used on Amiga systems) and is most commonly used on Apple Macintosh computer systems. The audio data in a standard AIFF file is uncompressed pulse-code modulation (PCM). There is also a compressed variant of AIFF known as AIFF-C or AIFC, with various defined compression codecs. Unlike the better-known lossy MP3 format, AIFF is uncompressed (which aids rapid streaming of multiple audio files from disk to the application), and is lossless. Like any uncompressed, lossless format, it uses much more disk space than MP3—about 10MB for one minute of stereo audio at a sample rate of 44.1 kHz and a bit depth of 16 bits. In addition to audio data, AIFF can include loop point data and the musical note of a sample, for use by hardware samplers and musical applications.

BWF: Broadcast Wave Format (BWF) is an extension of the popular Microsoft WAVE audio format and is the recording format of most file-based non-linear digital recorders used for motion picture, radio and television production. The purpose of this file format is the addition of metadata to facilitate the seamless exchange of sound data between different computer platforms and applications. It specifies the format of metadata, allowing audio processing elements to identify themselves, document their activities, and permits synchronization with other recordings. This metadata is stored as extension chunks in a standard digital audio WAV file.



Lossy Compression:
Lossy compression is a data reduction method that reduces the amount of data during the coding process, but retains enough information to be useful. For example, an MP3 file is a lossy music file that discards some of the original data but is still acceptable for music listening. The advantage of lossy compression is that the music file takes much less storage space. Lossy compression would be used when a small file size is the priority. It has adverse effects on the sound quality so would only be used in restrictive circumstances, such as mobile games.

MP3:
It is a common audio format for consumer audio streaming or storage, as well as a de facto standard of digital audio compression for the transfer and playback of music on most digital audio players. MP3 is a highly compressed format and has a very small file size, but the audio is of a lower quality. Because of the small file size you can easily download and email MP3s. There are numerous programs for playing MP3s available, such as Windows’ Media Player, Real Player, iTunes or WinAmp.

VOX: This is a file extension for Voxware software. It contains metadata that is used to stimulate human speech.VOX is based on the Dialogic ADPCM codec (Adaptive Differential Pulse Code Modulation). VOX files can be opened and edited using the VoxWare MetaSound codec. traditionally, files commonly have a sampling rate of 6000 or 8000 samples per second, but 8000 samples per second (8000 Hz) is more common. Unlike a WAV file, a VOX file does not contain a header to specify the encoding format or the sampling rate, so this information must be known in order to play the file.

Vorbis: is a free and open-source software project headed by the Xiph.Org Foundation (formerly Xiphophorus company). The project produces an audio coding format and software reference encoder/decoder (codec) for lossy audio compression. Vorbis is most commonly used in conjunction with the Ogg container format and it is therefore often referred to as Ogg Vorbis.

Audio Sampling

How can resolution and bit-depth constrain file size?

Bit depth: is the amount of bits(pieces of information) in an audio sample, it directly corresponds to the resolution. In general, the more bits that are available, the more accurate the resulting output from the data being processed. This also however means a larger file size. So the higher the bit rate the better the quality and the inverse is also true.

Sample Rate: is the number of samples of audio carried per second, measured in Hz or kHz (one kHz being 1 000 Hz). For example, 44 100 samples per second can be expressed as either 44 100 Hz, or 44.1 kHz.

Mono: Monaural or monophonic sound reproduction is single-channel. Typically there is only one microphone, one loudspeaker, or (in the case of headphones and multiple loudspeakers) channels are fed from a common signal path. In the case of multiple microphones the paths are mixed into a single signal path at some stage.

Monaural sound has been replaced by stereo sound in most entertainment applications. However, it remains the standard for radiotelephone communications, telephone networks, and audio induction loops for use with hearing aids.

Stereo: Stereophonic sound is a method of sound reproduction that creates an illusion of directionality and audible perspective. This is usually achieved by using two or more independent audio channels through a configuration of two or more loudspeakers (or stereo headphones) in such a way as to create the impression of sound heard from various directions, as in natural hearing.

Surround: Surround sound is a technique for enriching the sound reproduction quality of an audio source with additional audio channels from speakers that surround the listener (surround channels), providing sound from a 360° radius in the horizontal plane (2D) as opposed to "screen channels" (centre, [front] left, and [front] right) originating only from the listener's forward arc.

Surround sound is characterized by a listener location or sweet spot where the audio effects work best, and presents a fixed or forward perspective of the sound field to the listener at this location. The technique enhances the perception of sound spatialization by exploiting sound localization; a listener's ability to identify the location or origin of a detected sound in direction and distance. Typically this is achieved by using multiple discrete audio channels routed to an array of loudspeakers.

DSP – Digital Signal Processor

Digital signal processing (DSP) is the mathematical manipulation of an information signal to modify or improve it in some way. It is characterized by the representation of discrete time, discrete frequency, or other discrete domain signals by a sequence of numbers or symbols and the processing of these signals. The goal of DSP is usually to measure, filter and/or compress continuous real-world analog signals. Usually, the first step is conversion of the signal from an analog to a digital form, by sampling and then digitizing it using an analog-to-digital converter (ADC), which turns the analog signal into a stream of discrete digital values

Some limitations on recording audio are:

- Processing of signals involves more power consumption.

- Information is lost because we only take samples of the signal at intervals.

- Information may be lost when converting sound from analogue to digital.

RAM (pronounced ramm) is an acronym for random access memory, a type of computer memory that can be accessed randomly; that is, any byte of memory can be accessed without touching the preceding bytes. RAM is the most common type of memory found in computers and other devices, such as printers.

RAM can limit the recording and editing of sound, as complex software requires a lot of RAM to run smoothly. Video encoding and Sound encoding are very RAM and processor intensive. If the computer being used does not have enough RAM it may take much longer to encode the media.

Mp3 - It is the standard format for the recording and playback of mainstream music. It is a highly compressed file type so has a very small file size. This affects the quality of the recording so is not appropriate for media that requires high quality audio. In video games it may be suitable for a long music tracks to keep the file size small.

Wav - Wav is the main format used on Windows computers to store uncompressed, lossless audio. It is usually used in applications where file size is not a constraint. In video games, it is common to use Wav files for short sound effects.

Wav format is limited to files that are less than 4GB and some programs limit it further to 2GB. This translates to around 7 hours of CD quality audio. This is not appropriate for filming a feature length film, as the recording time could exceed this time and you may want higher quality audio.

Audio output (eg Mono, Stereo, Surround).

PCM – Pulse Code Modulation
In a brief sentence, pulse code modulation is a method used to convert an analog signal into a digital signal. So that it can be transmitted through a digital communication network, and then converted back into the original analog signal.

The PCM process includes three steps: Sampling, Quantization, and Coding.In a brief sentence, pulse code modulation is a method used to convert an analog signal into a digital signal. So that it can be transmitted through a digital communication network, and then converted back into the original analog signal.

The PCM process includes three steps: Sampling, Quantization, and Coding.

A PCM stream has two basic properties that determine the stream's fidelity to the original analog signal: the sampling rate, which is the number of times per second that samples are taken; and the bit depth, which determines the number of possible digital values that can be used to represent each sample.

In what types of scenario may you use the following audio recording equipment?

For example

Multitrack recording (MTR)—also known as multitracking, double tracking, or tracking—is a method of sound recording that allows for the separate recording of multiple sound sources to create a cohesive whole. Multitrack recording is a process where the tape is divided into multiple tracks parallel with each other. Because they are carried on the same medium, the tracks stay in perfect synchronisation. The first development in multitracking was stereo sound, which divided the recording head into two tracks. Stereo sound has become the standard format in most media and so Multi-track recording is often used.

 Midi – Multi Instrument Interface - is information that computers use to play back certain instrument sounds. It works on any computer hardware that utilises a synthesiser. For example: when you record music onto a computer using MIDI, the software saves this list of messages and instructions as a .MID file. If you play the .MID file back on an electronic keyboard, the keyboard's internal synthesizer software follows the instructions to play back the song. The keyboard will play a certain key with a certain velocity and hold it for a specified amount of time before moving on to the next note. Therefore MIDI is used in a variety of areas to create or recreate music without the use of the original instruments.

 DAT - Digital Audio Tape (DAT or R-DAT) is a signal recording and playback medium developed by Sony and introduced in 1987. In appearance it is similar to a Compact Cassette, using 4 mm magnetic tape enclosed in a protective shell, but is roughly half the size at 73 mm × 54 mm × 10.5 mm. As the name suggests, the recording is digital rather than analog. DAT has the ability to record at higher, equal or lower sampling rates than a CD (48, 44.1 or 32 kHz sampling rate respectively) at 16 bits quantization. If a digital source is copied then the DAT will produce an exact clone, unlike other digital media such as Digital Compact Cassette or non-Hi-MDMiniDisc, both of which use a lossy data reduction system
  Analogue Software Plug-ins
Analogue - relating to or using signals or information represented by a continuously variable physical quantity such as spatial position, voltage, etc.
 Software Sequencer - A music sequencer (or simply sequencer) is a device or application software that can record, edit, or play back music, by handling note and performance information in several forms, typically MIDI or CV/Gate, and possibly audio and automation data for DAWs and plug-ins.

Music sequencers are often categorized by handling data types, as following:

• MIDI data on the MIDI sequencers (implemented as hardware or software)
• CV/Gate data on the analog sequencers, and possibly others (via CV/Gate interfaces).
• Automation data for DAWs and plug-ins.

  on the DAW with sequencing features, and software instrument/effect plug-ins on them.

• Audio data

  on the audio sequencers including DAW, loop-based music software, etc.;
  or, the phrase samplers including Groove machines, etc.