VOCODER, COMPANDER, Etc.

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While voice frequencies are linear or sequential, human voice has unique characteristics. That is, quantization levels according to a logarithmic, instead of a linear, process provides smaller quantization levels at lower signal often lower frequency - bass voice. The result is a higher level of reproduction for human voice.

A VOCODER [VOice-CODER] device allows the DSP [Digital Signal Processor] to amplify (quantize) lower or higher frequencies based on the application. That is, the VOCODER can change the logarithms for processing the speech or other input signal such as music. Such a device is also called a COMPANDER [COMpressor and exPANDer].

In other words, different Companders will process voice samples differently. The North American PCM [Pulse Code Modulation] is called µ-Law (µ from the Greek mu or M or Modulation), which approximates the logarithmic (steps) curve.

Mu-Law encoding is where the steps within a cord are all the same size and therefore linear (the art approximates this but is not exact).
Analog samples are placed into digital quantities (quantizing) at specific intervals which means logarithmic (step) quantizing is not a linear process or not all analog values are sampled. However, there are enough samples to reproduce voice but not hi-fidelity sounds. That is, different quantizing systems such as µ-Law and A-Law sample voice differently resulting in different reproduction qualities.

In a PCM [Pulse Code Modulation] sample of 8-bits, there are 256 possible quantized digital values. PCM assigns these 256 possible values as 127 positive and 127 negative encoding levels. Each of the two higher levels are divided into 8 separate levels (bands) called Chords.

Coding-encoding is the process of sampling quantities and putting them into digital values of voice, music or other sounds. The Nyquist-Shannon sampling theorem states that the sampling frequency must be at least twice as high as the highest input frequency for the result to closely resemble the original signal. A 4,000 Hz voice pattern would be sampled at 8,000 BPS [Bits Per Second]. Organized into 24 separate voice channels with spacing bits (called Framing bits) separating each 24 segments of 8,000 bits becomes a T-1-Transmission Level One transmission circuit of 1,544,000 BPS. For example, in MPEG3 [Motion Picture Experts Groups version 3], different compression (sampling or quantizing) rates are needed for different music quality levels such as 128 KBPS - CD quality (twice normal bandwidth), 96 KBPS - near-CD quality and 64 KBPS - FM radio quality.

Analog-to-digital conversion is performed by a special microprocessor and memory device called a DSP. The function of the DSP [Digital Signal Processor/Processing] is to convert analog signals into digital values (ones & zeros) for transmission. A DSP may also perform other functions such as encapsulating digital transmission (1s/0s) into packets such as RTP [Realtime Transport Protocol], Ethernet, ATM [Asynchronous Transfer Mode] or other data formats.

Voice Frame Size in MSEC or voice sample size forces the VoIP (using RTP) designer to balance voice quality with potential network delays which could cause jitter or voice packet loss. Various CODECs sample at 1 msec (millisecond - 1/1000), 10, 20 and 30 msec. The larger the voice sample, the better the quality however the greater the potential delay called latency from other network voice and/or data packets causing the voice to jitter.

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