In digital audio, the purpose of binary numbers is to express the values of samples that represent analog sound. (contrasted to MIDI binary use)
characteristics of sound ANALOG DIGITAL frequency (time based) sample rate (time based) = measured in hertz - samples per second measured in hertz - cycles per second
characteristics of sound ANALOG DIGITAL sample rate (time) frequency (time) = quantization(level) amplitude (level) = measured in bits measured in decibel
characteristics of sound analog-to-digital conversion : two steps (1) sampling (2) quantization
sampling theorem • “A continuous audio signal can be replaced by a discrete sequence of samples without loss of any information and the original continuous audio signal can be reconstructed from the samples.”
When a digital recorder takes a sample, it takes a snapshot of the audio waveform and turns it into bits that can be stored and manipulated. 010001111101100000110010011000001010010010100011111011000001100100110000010100100101000111110110000011001001100000101001001010001111101100000110010011000001010010010100011111011000001100100110000010100100101000111110110000011001001100000101001001010001111101100000110010011000001010010010100011111011000001100100110000010100100101000111110110000011001001100000101001001010001111101100000110010011000001010010010100011111011000001100100110000010100100101000111110110000011001001100000101001001 0100011111011000001100100110000010100100101000111110110000011001001100000101001001
sample rate The frequency of the 'snapshots' of the audio stream in a single second Just as in measuring frequency, hertz is used to define the number of samples taken per second
flip cards http://www.youtube.com/watch?v=xSrDnIVgVv0
flip cards http://www.youtube.com/watch?v=Ql4Kkb-89D0
sample rate: the number of samples (measurements) taken of an analog signal in 1 second
sample rate: The sample rate determines the frequency range (bandwidth) of a system.
sample rate: the number of samples (measurements) taken of an analog signal signal in 1 second The sample rate determines the frequency range (bandwidth) of a system. The faster the sample rate, the better the accuracy of getting a true picture of higher frequencies.
Some common sample rates are: 22,050 aka 22.05 kHz - 22,050 samples per second. A sample every 1/22,050 of a sec. 24,000 aka 24 kHz - 24,000 samples per second. A sample every 1/24,000 of a sec. 30,000 aka 30 kHz - 30,000 samples per second. A sample every 1/30,000 of a sec. 44,100 aka 44.1 kHz - 44,100 samples per second. A sample every 1/44,000 of a sec. 48,000 aka 48 kHz - 48,000 samples per second. A sample every 1/48,000 of a sec.
The higher the sample rate, the better the quality of the sample. A sample taken at 44.1 kHz will contain twice the information as a sample taken at 22,050 kHz.
High sample rates are better at capturing high frequency waveforms, but if you are sampling lower frequency sounds, such as kick drum, bass, etc.,...not as critical. you might consider sampling at the lower rate to save hard drive space.
NYQUIST THEORY Named after a Bell engineer who worked on the speed of telegraphs in the 1920s
There must be two samples per period. In other words, the sampling frequency must be at least twice the highest signal frequency recorded in order to be effective.
Sample rates with Nyquist yield 22,050 kHz = 11,025 kHz (Nyquist) 24,000 kHz = 12,000 kHz 30,000 kHz = 15,000 kHz 44,100 kHz = 22,050 kHz
It is therefore important to take into consideration the highest frequency of the audio material to be recorded.
If a frequency of A-14,080 Hz is to be recorded, a sample rate of 44.1 kHz would be the logical choice to use. 14,080 Hz falls within the range of the Nyquist of 44.1 kHz which is 22.05 kHz.
CD is 44.1 kHz The choice of sample rate determines the audio bandwidth of the recorder used. Considering that the human hearing range at best ranges from 20 Hz to 20 kHz, a 44.1 kHz sample rate theoretically should satisfy most audio needs.
If a 25 kHz waveform is sampled at 44.1 kHz (which has a Nyquist value of 22.05 kHz), the Nyquist rule is broken. 44 kHz - 25 kHz , results in a 19 kHz waveform which is heard as distortion. This is also known as aliasing or foldover.
- 2k - 4k In audio: Alias is mirrored the same distance below the Nyquist frequency as the original was above it, at the original amplitude (foldover)
Once aliasing is introduced into the digital stream, it can’t be removed. It must be stopped before entering the digital stream.
QUANTIZATION The technique of measuring an audio event to form a numerical value. In digital audio, the values = voltages voltages represent amplitude the more bits, the more voltages that can be represented.
sample rate = frequency quantization = amplitude gain staging (p. 133)
quantization error The difference between the actual analog value at the sample time and the selected quantization interval value
The amplitude of the audio signal is broken down into a series of discrete steps. Each step is then given a binary word that digitally encodes the level of the signal. The length of the digital word determines the quality of the representation.
etc 0111 1100 0110 0011 1000 0100 0010 0001 0000 4 bits 16 voltages
The larger the word, the better the quality (16 bit word compared to an 8 bit word). The larger the bit word, the greater the headroom of the audio system (6 dB for every bit).
The more steps with which to describe the signal, the smoother the result will be.
SAMPLE BITS The more bits used to describe something, the better the clarity and fidelity An 8 bit sample contains 256 steps of information while a 16 bit sample contains up to 65,536 steps.
The bit resolution of a system defines the dynamic range of the system. 6dB is gained for every bit signal to noise ratio (analog) signal to error ratio (digital)
8 bits equals 256 states = 48 dB 16 bits equals 65,536 states = 96 dB To find the dynamic range of a system, multiply the bit rate X 6.
In a 16 bit system, there are 65,536 different numbers, each number representing a different analog signal voltage