Samplers

How do they work?

Sampling

ADC

Sounds from the real world can be recorded and digitized using an Analogto-Digital Converter (ADC). As in the diagram below, the circuit takes a sample of the instantaneous amplitude (not frequency) of the analogue waveform. Frequencies will be recreated later by playing back the sequential sample amplitudes at a specified rate

Sample Rate

Samples are taken at a regular time intervals called the sampling rate. According to the Nyquist theorem (named after Harry Nyquist), the highest reproducible frequency of a digital system is 1/2 the sampling rate, often called the Nyquist frequency. A sampling rate of 44,100 samples per second, the rate at which CD's are encoded, can reproduce frequencies up to 22,050 Hz, well above the 20,000 Hz limit of human hearing. Frequencies that are recorded above the Nyquist frequency may foldover at a much lower frequency than the original, which is called aliasing.

Aliasing

If the signal to be sampled contains frequencies that exceed one-half the sampling rate, these frequencies will be sampled but will appear to be lower in frequency when outputted. This phenomenon, called foldback or aliasing, occurs where rogue frequencies appear in the signal. These rogue frequencies are related to amount by which the signal's frequency exceeds half sampling frequency.

Aliasing

Anti Aliasing Filter

To avoid aliasing, a steep (brickwall) filter is used. This filter is placed before the ADC input to prevent signals above the Nyquist frequency from entering the system. Standard sampling rates are 44.1K are 48K (and even 96K in some high-end recording systems).

Anti Aliasing Filter

In essence the ANTI ALIASING FILTER is a Low Pass Filter if the sampling rate is 44.1kHz then the filter is used to remove any frequency above 22.1kHz, well above the frequency of normal hearing.

ADC

The ADC 'measures' the instantaneous amplitude of the input signal at regular intervals - for CD quality it measures the amplitude 44100 times per second. These measurements are then represented as a series of 1 and 0s. The size of number which can represent these measurements is determined by the bit rate. The higher the bit rate the higher the quality and greater the dynamic range

Bit Rate

The number of available values is determined by the number of bits (0's and 1's) used for each sample in a process called quantization. When a sample is quantized, the analogue amplitude has to be rounded off to the nearest available digital value. This rounding-off process is called approximation. The smaller the number of bits used per sample, the greater distances the analogue values need to be rounded off to. The difference between the analogue value and the digital value is called the approximation or quantizing error

Bit Rate

Bit Rate

The greater the approximation error, the greater the amount of digital or quantizing noise produced. 6 dB of dynamic range for every additional bit used per sample. The CD/DAT standard 16-bit samples, with their impressive 65,536 values for quantizing, provide the theoretical playback system optimum of 96 dB dynamic range.

Bit Rate

CD quality is 16 bit this means that each discrete sample measurement can be represented in 65,500 value. The relationship between these values and the bit rate is given by D=2n where D is the available number of values and n is the bit rate. We can see the that the quality of sampling/dynamic range is not a linear relationship so 16 bit is not twice as good as 8bit but many times better 8 bit gives us 256 values for example - image colours on your screen 16 bit 65,500 20 bit =2 20

Bit Rate and Dynamic Range

The relationship between bit rate and dynamic range is given by S=6ndB so 8 bit is 48bB 16 bit is 96dB 20 bit = 120dB

Memory

Once the signal has been sampled and converted into a series of 1s and 0s it is then stored in the samplers/computers memory RAM Random Access memory -this is the computers/samplers memory area where the temporary storing and editing of samples occurs Typical editing functions are topping and tailing, looping, reversing, time stretching etc occur

Editing

Editing Topping and tailing - remove noise unwanted sections etc Looping Used to elongate sounds - a section at the end of a sample would be looped indefinitely to produce sustain - save memory The transition from the end of the sample to the loop point is critical as these is where any differences in amplitudes etc will he heard. If not correct then click etc are heard, this is where we use xfades to even out the velocities at the beginning and end of loops. Like a loop of tape Time stretching Changing the pitch keeping the time the same changing the time, keeping the pitch the same reversing - as it says

Multi sampling

Take many samples from an instrument to reduce the phenomenon called munchkinisation. If a single note on an instrument is sampled and put across the keyboard then as we move up/down the keyboard from the original pitch of the sample, it sounds less and less like the original instrument. To over come this instruments are multi samples. The samples can be velocity triggered to provide subtle nuances of the instrument. As the trumpet plays louder it gets brighter and so on.

Storage and Types

Originally samples were stored on 51/2 floppy disks each disc could only hold a megabit 31/2 inch floppy disks could hole just over 1meg Then zip drives, hard discs etc Now stored on computer hard drives.

DAC

DAC - this is where the signal is converted back into an analogue signal i.e. a continuously varying signal/voltage as opposed to a discrete digital signal. Very simply the DAC emits a voltage proportional to the number inputted. This creates a stair step reconstruction of the original signal. A reconstruction filter (LPF) is used to smooth the transitions of the stair step signal.

Block Diagram of a Sampler

Brick Wall filter Anti Aliasing Filter Does not allow any frequency above 20kHz to pass through

ADC

Analogue to digital convertor

Random Access Memory Digital info Stored in memory where it can be manipulated

DAC

LPF Reconstruction Filter

Digital to analogue converter

Types

Stand alone - Akai Keyboard Computer- based - Logic's EXS24