15 - Advanced Digital Sound

 1. Mickey is a recording artist. He inputs his voice into a computer and it is converted into a ____________________ using an ADC.
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

  PNG format file

  music file

  digital data file

  analogue file

 2. Read the (long!) excerpt below and fill in the blanks at the end.
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 ___________

  analogue rate

  sampling rate

  digital sound rate

  bandwidth rate

 3. To create digital sound as close to the real thing as possible you need to take ____________________ as you can. When recording MP3s you'll normally use a sampling rate between 32,000, 44,100 and 48,000 Hz (samples per second)

   as many samples per second

  a maximum of two samples

  as few samples as possible to increase quality

  None of the above

 4. The digital sound is 100% exactly the same shape as the analogue wave and therefore the sound quality is identical.



 5. The sound quality can be improved by decreasing the sample rate



 6. Sampling rate (measured in Hz) is __________

  the number of bits used to store the analogue input per minute

  None of the above

  the number of samples taken per millisecond from the digital input

  the number of samples taken per second from the analogue input to create a digital signal

 7. Some suggest that the sound must be sampled at twice the highest analogue input frequency to create an accurate representation of the original input waveform. This is called:

  Nyquist's theorem

  Music doubling

  lowering the sample rate

  Michael Jackson's theorem

 8. Sampling resolution is also called audio bit depth and is the _______________

  number of samples used to store each bit

  number of bits used to store each sample

  None of the above

  number of pixels used to store the sound

 9. How much disk space would a 120 second sound recording require?
Sample frequency = 4000Hz, Sample Resolution = 16 bits (2 bytes)

  None of the above

  File size (bytes) = sample resolution x length of sound x frequency of bits = 4000 x 2 x 90 = 3234232

  File size(bytes) = sample frequency x sample resolution x length of sound. 4000 x 2 x 120 = 960,000 bytes

  There is not enough information to make this calculation

 10. If you wanted to record a 30 second voice message on your mobile phone you would use the following. Is the calculation at the end correct (true) or incorrect (false)?
Sample Rate = 8,000 Hz
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



 11. What would this give you? Sampling Rate * Sampling Resolution

  the sampling rate x 2

  the bit rate of a song

  the music resolution rate of a song

  None of the above

 12. Work out the sample rate of the following sound file:
Sound File = 100,000 bits
Sample Resolution = 10 bit
Length of Sound = 5 seconds

  100,000 x 10 x 5 = 5000000000

  None of the above

  100,000 / 5 =20,000 Hz

  100,000 / (10 * 5) = 2,000 Hz

 13. Why might a song recorded with the following settings have a file size of 7,040,000 bits?
Sample Rate = 22,000 Hz
Sample Resolution = 16 bit
Length of Sound = 10 seconds

  The file might be recorded in stereo, meaning twice the amount of data would have to be stored

  The file might be corrupted

  The file might be recorded in mono - meaning only half the amount of data would need storing

  None of the above

 14. Telephone networks and VOIP services can use a sample rate as low as 8 kHz. This uses less data to represent the audio



 15. At 8 kHz, the human voice can still be heard clearly - but music at this sample rate would sound low quality.