Voice digitization scheme that samples an analog voice signal 8000 times per second and converts each sample to an 8-bit code, yielding a digital voice rate of 64 kbps, allowing 30 timeslots on a 2-Mbps line; defined in ITU (ex-CCITT) Recommendation G.711.
Common form of transferring analog information into digital signals by representing analog waveforms with a stream of digital bits forming words that relate the amplitude of a signal at a certain point (the sample). The word length used, the number of bits used to represent the amplitude of a sample is a determinant in the quality of reproduction along with the sampling rate (the number of samples taken per second). The word length used for standard CDs is 16 bits meaning that each amplitude is represented by a string of 16 ones and zeroes. Higher quality formats use 20 bit and 24 bit word lengths. A 20 bit word breaks up the height into 1,048,576 pieces, about 16 times more resolution that a 16 bit word (with 65,553 pieces). A 24-bit word breaks the amplitude up into 16,777,216 pieces, about 256 times the information of a 16-bit word.
Method of sampling information signals at regular intervals and transmitting the samples as a series of pulses in coded form which represent the amplitude of the information signal at that time.
A method for converting analog voice to digital voice. The analog voice signal is sampled at a rate of 8000 times per second and each sample is described digitally using an 8-bit byte. This results in a 64 kbps digital voice signal. (source)
A technique for digitizing audio into an uncompressed format by assigning a value to the amplitude of the signal at fixed intervals. RGB A color model that describes color information in terms of the red (R), green (G), and blue (B) intensities that make up the color. sampling rate The frequency of sampling. The higher the sampling rate (that is, the more samples taken per unit of time), the more closely the digitized result resembles the original.
This is a time division modulation technique in which analog signals are transformed into quantized digital signals.
The internationally accepted Codex used by telephone companies to translate between the 56 and 64Kbps digital signaling technologies and the analog signals sent across POTS telephone lines. PCM codes are seven or eight bits in size, meaning each code byte has 128 or 256 possible values. (North American POTS connections generally only use 7 bit codes.) Or a commonly employed algorithm to digitize an analog signal (such as a human voice) into a digital bit stream using simple analog to digital conversion techniques.
The process that samples quantizes and codes the modulating analog signal into digital bit stream.
A method of representing an audio signal as a series of digital samples. Circuitry found in every audio and video component that converts 60 Hz alternating current from the wall outlet into direct current that supplies the device's circuitry.
An uncompressed wave data format in which each value represents the amplitude of the signal at the time of sampling. registered parameter number (RPN) An identification number for a registered parameter that has been assigned a function by the MIDI Manufacturers Association and can be accessed through controllers.
a basic form of digital modulation where an analog signal is sampled with each sample being quantized independently of the other samples.
A method used to convert an analog signal into noise-free digital data that can be stored and manipulated by computer. PCM takes an 8-bit sample of a 4kHz bandwidth 8000 times a second, which gives 16K of data per second.
A common method for digitizing voice signals. The bandwidth required for a single digitized voice channel is 64 kilobits per second.
one method of digitizing an analog picture.
A modulation technique in which the signal being transmitted is sampled at regular intervals to determine its magnitude. The magnitude is converted to a digital pulse for transmission.
A process in which a signal is sampled, and the magnitude of each sample with respect to a fixed reference is quantized and converted by coding to a digital signal.
The most common way of converting an analogue source into a digital form. This works by taking samples of the continuously varying analogue signal at regular intervals. At each sampling point a number is generated to represent the size of the signal.
The effect of sampling an analog signal.
Method of modulation in which signals are sampled and converted to digital words that are then transmitted serially. Most PCM systems use either 7- or 8-bit binary codes. There are, however, several standards for PCM coding: most common are ?-Law in North America and A-Law in Europe (both based on logarithmic conversion of the signal).
A common form of encoding and transmission in which analogue speech signals are converted to digital format for the purpose of multiplexing and transmission over distance. PCM is commonly used in multi-circuit inter-exchange trunk system, or "subscriber carrier" systems in which a number of telephone subscribers along a route are served from the same cable.
The most frequently used method of convening analog signals into digital bits.
(PCM) a way to convert sound or analog information to binary information (0s and 1s) by taking samples of the sound and record the resulting number as binary information. Used on all CDs, DVD-Audio, and just about every other digital audio format. It can sometimes be found on DVD-Video.
A digital modulation method of encoding voice signals into digital signals. Also called PCM.
This is a binary digital signal format used for digitizing analog data.
A method of quantizing audio-range analog signals into a digital form for transmission in digital communications systems or for processing in DSP. Effectively the same as Analog-to-Digital conversion.
A generic term for digital transmission.
PCM): A sophisticated technique for reducing voice data storage requirements that is used by Dialogic in the voice boards. Dialogic supports either _-law Pulse Code Modulation, which is used in North America and Japan, or A-law Pulse Code Modulation, which is used in the rest of the world.
A technique used in DSP voice boards for reducing voice data storage requirements. Dialogic supports either mu-law PCM, which is used in North America and Japan, or A-law PCM, which is used in the rest of the world.
A time division modulation technique in which analog signals are sampled and quantized at periodic intervals into digital signals. The values observed are typically represented by a coded arrangement of 8 bits of which one may be for parity.
A sampling technique for digitizing analogue signals, especially audio signals. When you encode audio by using PCM, no compression is applied.
A technique for converting an analog signal with an infinite number of possible values into discrete binary digital words that have a finite number of values. The waveform is sampled, then the sample is quantized into PCM codes.