1.1.3 Sound
1.1.3 Sound
1.1.3 Sound
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sound particles will collide with your ear drum, vibrating it and sending a message to your brain. This is
how you hear:
When you hear different volumes and pitches of sound all that is happening is that each sound wave
varies in energy for the volume (larger energy waves, the louder the sound), or distance between sound
waves which adjusts the pitch, (smaller distances between waves leads to higher pitched sound).
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A computer representation of a stereo song, if you look carefully you'll see the volume of the song varying
as you go through it
This section of the book will cover how we record, store and transmit sound using computers. Sound
waves in nature are continuous; this means they have an almost infinite amount of detail that you could
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store for even the shortest sound. This makes them very difficult to record perfectly, as computers can
only store discrete data, data that has a limited number of data points.
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An analogue sound wave is picked up by a microphone and sent to an Analogue to Digital (ADC) converter
in the form of analogue electrical signals. The ADC converts the electrical signals into digital values which
can be stored on a computer.
Once in a digital format you can edit sounds with programs such as audacity.
To play digital audio you convert the sound from digital values into analogue electrical signals using the
DAC, these signals are then passed to a speaker that vibrating the speaker cone, moving the air to create
sound waves and analogue noise.
Analogue to Digital Converter (ADC) - Converts analogue sound into digital signals that can be stored on a
computer
Digital to Analogue Converter (DAC) - Converts digital signals stored on a computer into analogue sound
that can be played through devices such as speakers
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Hertz (Hz) - the SI unit of frequency defined as the number of cycles per second of a periodic
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phenomenon
To create digital music that sounds close to the real thing you need to look at the analogue sound waves
and try to represent them digitally. This requires you to try to replicate the analogue (and continuous)
waves as discrete values. The first step in doing this is deciding how often you should sample the sound
wave, if you do it too little, the sample stored on a computer will sound very distant from the one being
recorded. Sample too often and sound stored will resemble that being recorded but having to store each
of the samples means you'll get very large file sizes. To decide how often you are going to sample the
analogue signal is called the sampling rate. Take a look at the following example:
Original Sound
digital loses
sharpness
loss of peaks
poor resemblance
to original!
To create digital sound as close to the real thing as possible you need to take as many samples per
second as you can. When recording MP3s you'll normally use a sampling rate between 32,000, 44,100 and
48,000Hz (samples per second). That means that for a sampling rate of 44,100, sound waves will have
been sampled 44,100 times per second! Recording the human voice requires a lower sampling rate,
around 8,000Hz.
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Comparison of the same sound sample recorded at 8kHz, 22kHz and 44kHz sample rate. Note the spacing
of the data points for each sample. The higher the sample rate the more data points we'll need to store
Sampling resolution
Sampling resolution - the number of bits assigned to each sample
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As you saw earlier, different sounds can have different volumes. The sampling resolution allows you to set
the range of volumes storable for each sample. If you have a low sampling resolution then the range of
volumes will be very limited, if you have a high sampling resolution then the file size may become
unfeasible. The sampling resolution for a CD is 16 bits used per sample.
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To work out the size of a sound sample requires the following equation:
File Size = Sample Rate * Sample Resolution * Length of sound
This is the same as saying:
File Size = Bit Rate * Length of sound
Let's look at an example:
Example: Sound File Sizes
If you wanted to record a 30 second voice message on your mobile phone you would use the following:
Sample Rate = 8,000Hz
Sample Resolution = 16 bit
Length of Sound = 30 seconds
Therefore the total file size would be:
8,000 * 16 * 30 = 3 840 000 Bits = 480 000 Bytes
Sound Editing
If you are interested in sound editing you can start editing your own music using a program
called Audacity. Using Audacity you can create your own sound samples with different sample rates and
sample resolutions, listening to the difference between them and noting the different file sizes.
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Features
This is a list of features in Audacity, the free, open source, cross-platform audio editor.
Recording
Audacity can record live audio through a microphone or mixer, or digitize recordings from cassette tapes,
records or minidiscs. With some sound cards, and on any Windows Vista, Windows 7 or Windows 8
machine, Audacity can also capture streaming audio.
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Sound Quality
Supports 16-bit, 24-bit and 32-bit (floating point) samples (the latter preserves samples in excess
of full scale).
Sample rates and formats are converted using high-quality resampling and dithering.
Tracks with different sample rates or formats are converted automatically in real time.
Editing
Easy editing with Cut, Copy, Paste and Delete.
Unlimited sequential Undo (and Redo) to go back any number of steps.
Edit and mix large numbers of tracks.
Multiple clips are allowed per track.
Label tracks with selectable Sync-Lock Tracks feature for keeping tracks and labels synchronized.
Draw Tool to alter individual sample points.
Envelope Tool to fade the volume up or down smoothly.
Automatic Crash Recovery in the event of abnormal program termination.
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Accessibility
Tracks and selections can be fully manipulated using the keyboard.
Large range of keyboard shortcuts.
Excellent support for JAWS, NVDA and other screen readers on Windows, and for VoiceOver on
Mac.
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Effects
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Plug-ins
Support for LADSPA, Nyquist, VST and Audio Unit effect plug-ins.
Effects written in the Nyquist programming language can be easily modified in a text editor - or
you can even write your own plug-in.
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Analysis
Spectrogram view modes for visualizing frequencies.
"Plot Spectrum" command for detailed frequency analysis.
"Sample Data Export" for exporting a file containing amplitude values for each sample in the
selection.
Contrast Analysis for analyzing average rms volume differences between foreground speech and
background music.
Support for adding VAMP analysis plug-ins.
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