I Power

Higher Computing

Multimedia technology

Audio

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Sound cards Sound cards carry out the following jobs:

recording audio, playback of digitised audio, playback of audio CDs, sound synthesis, interfacing with MIDI instruments, digital input and output for transferring files.

To help them carry out these tasks, sound cards have ADCs and DACs as well as DSPs.

Sound cards use the following techniques for capturing sound data.

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Analogue to Digital Convertors (ADCs) The analogue signals from the array of

CCDs is fed to analogue to digital convertors. These ADCs receive a continuous streams of analogue current which they convert to digital data representing the sound.

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Digital to Analogue Convertors (DACs) The digital to analogue converter (DAC) takes the

digital data encoding the sound and changes it into a varying analogue signal.

This is fed out through the sound line out socket and is used to control the diaphragm in the speaker which creates the sound waves you hear.

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Pulse Code Modulation (PCM)

A method of encoding information in a signal by varying the amplitude of the pulses. This limits pulse amplitude to several predefined values.

This technique is used by codecs to convert an analogue signal into a digital bit stream. The amplitude of the analogue signal is sampled and converted into a digital value: It is described as raw because the digitised data has not been processed further, for example by compressing it. Raw PCM sound files can be very large indeed.

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PCM (Cont.)

The original signal is given.

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PCM (Cont.)

This sound wave is then sampled at regular intervals.

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PCM (Cont.)

The signal is then split into different levels.

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PCM (Cont.)

Each Pulse level is then recorded giving us the digital signal.

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Bit rate This term is used to describe the number of

bits that are sent in one second to transmit a sound file. Stereo CD needs a bit rate of 1378 kbps and an MP3 file 384 kbps.

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Adaptive Delta Pulse Code Modulation (ADPCM) This compresses data that has been encoded in PCM

form. It stores only the changes between the samples, not the samples themselves. This compresses PCM data by a ratio of 4:1 since it uses only 4 bits for the sample change rather than the 16 bits for the original PCM value.

Microsoft WAV format uses ADPCM. This means that many Windows programs can play WAV files using the Windows sound driver. WAV is the standard for storing sound files on Windows systems and can be sampled at a bit depth of either 8 or 16 bits and one of the following sampling rates 11.025 kHz, 22.05 kHz or 44.1 kHz. WAV files can be very large. One minute of sound can take up as much as 27 MB of storage.

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ADPCM (Cont.)

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Resource Interchange File Format (RIFF) This is a file format for multimedia data on

PCs. It can contain bit-mapped graphics, animation, digital audio and MIDI data.

The WAV file format is the RIFF format for storing sound data.

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MP3 Its full title is MPEG-1/2 Layer 3. It is a format for

compressing sounds which uses a lossy technique that does not seriously degrade the quality of the sound because it filters out aspects of the original sound that the human ear cannot detect. After filtering it then applies further compression techniques. A form of coding called Huffman encoding is used to compress the data once it has been captured.

One minute of music takes up around 1 MS of space. MP3 allows compression of CD quality audio files by a factor of 12 with little loss in quality. This explains why it is such a widely used format.

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MIDI The Musical Instrument Digital Interface

(MIDI) is a standard interface used by musical instruments like keyboards, synthesisers and drum machines which enables notes played on an instrument to be saved on a computer system, edited and played back through a MIDI device.

The information about the sound is stored in a MIDI file which the computer can then use to tell the instrument which notes to play.

http://www.cool-midi.com/

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MIDI (Cont.) When a MIDI sound is stored in a computer system the

following attributes or properties of the sound are stored: Instrument, Pitch, Volume, Duration Defines the instrument being played. Each built-in sound on a

MIDI keyboard has an instrument number assigned to it. When selected the instrument number is saved by the computer so that, on playback, the notes in the musical piece are played with the sound of that specific instrument.

This sets the musical tone of a note which is determined by the frequency.

This controls the loudness or amplitude of the note. This determines the length of a note (the number of beats). Tempo Rate is the speed at which the piece of music is set.

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MIDI (Cont.) Advantages of MIDI Allows musical pieces or messages to be exchanged and edited on different

computers. It is an easily manipulated form of data. Changing the tempo is a

straightforward matter of changing one of the attributes. A musician can store the messages generated by many instruments in one

file. This enables a musician to put together and edit a piece of music generated on different midi instruments with complete control over each note of each instrument.

Produces much smaller file sizes than other sound formats. Because it is digital it is easy to interface instruments, such as keyboards, to

computers. The musician can store music on the computer and the computer can then play the music back on the instrument.

Disadvantage Browsers require separate plug-ins to play MIDI files.

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Normalising sound files When sound files are sampled, different sounds are louder

than others. Background noises or voices might be too loud or too quiet. Different music tracks might playback at different levels. To avoid this sound files are normalised.

This means that the signal levels are adjusted so that they all fall into line with the average volume of all of the sounds on the recording. The normalising function on sound editing software scans the uncompressed audio file to determine the peak or average level and increases or decreases the levels throughout the file to obtain the desired volume level.

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Clipping Sound Files We have all listened to sound files that do not sound

good. Part of the sound seems unclear or missing. The most probable cause of this is clipping.

If a sound is recorded at too high a level then the sound wave will be automatically clipped. This means that the top of the sound wave is cut off.

Some sound editing software will indicate to the user which amplitudes in a recording are being dipped and offer the option of reducing the recording volume.

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Clipping Sound Files

Sound wave before amplification

The Same sound wave after amplification

Here Clipping has occurred.

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Fade This means to gradually reduce the recording

volume of a sound so that it dies away slowly. Many sound editors give the user graphical

controls which they can use to control the length of the fadeout and the rate at which the volume drops.

Most sound editing software comes provided with fade settings, sometimes called 'envelopes' and also lets the user define and save their own 'envelopes'.

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Fade

Sample of music

Sample of music which has been faded.

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Monophonic Sound Commonly called mono sound, mono, or

non-stereo sound, this early sound system used a single channel of audio for sound output. In monophonic sound systems, the signal sent to the sound system encodes one single stream of sound and it usually uses just one speaker. Monophonic sound is the most basic format of sound output.

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Stereophonic Sound This means the audio is recorded on two

sound channels using a separate microphone for each channel. Sounds nearest the left microphone will record loudest on the left channel, similarly for the right channel.

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Surround sound This uses speakers to surround the

listener with a circle of sound. The Dolby Surround Pro Logic system uses two speakers in front and two behind.

• This distinguishes between the original sound and the listener's perception of that sound in different environments and from different directions.

• This creates the illusion of sound coming from a specific location or reflecting off different surfaces.

• The software uses algorithms to create an 'all round' sound effect. It is based on a mathematical filter which is applied to the sound data.

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Digital signal processor (DSP) The DSP is an integrated circuit designed for high

speed data manipulation used in image manipulation (as well as audio and other applications, for example communications).

The DSP's main function is to compress and decompress sound files as well as provide enhancements to sounds, for example reverberation.

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Calculating the size of a sound file We use the following formula to calculate

the size of a sound file

File size (B) =

Sampling frequency (Hz) x

Sound time (s) x

Sampling depth (B) x

Channels

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Calculating the size of a sound file (Cont.)

Use this formula to calculate the file size of 1 minute of mono sound sampled at a frequency of 22.05 kHz and a bit depth of 8 bits.

File size = 22.05 x 103

sampling freq(Hz)

x 60

Time(s)

x1

Sampling Depth (Bytes)

x1

No of Channels

= 1 323 000 bytes = 1.26 MB