|
|||||||||||||||||||||||||||||||||
|
|||||||||||||||||||||||||||||||||
In digital recording, the analog signal of video or sound is converted into a stream of discrete numbers, representing the changes in air pressure or chroma and luminance values through time; thus making an abstract template for the original sound or moving image.
History
ProcessRecording
Playback
Getting the bits recordedEven after getting the signal converted to bits, it is still difficult to record: the hardest part is finding a scheme that can record the bits fast enough to keep up with the signal. For example, to record two channels of audio at 44.1 kHz sample rate with a 16 bit word size, the recording software has to handle 1,411,200 bits per second. Techniques to record to commercial mediaFor digital cassettes, the read/write head moves as well as the tape in order to maintain a high enough speed to keep the bits at a manageable size. For CDs or DVDs, a laser is used to burn microscopic holes into the dye layer of the medium. A weaker laser is used to read these signals. This works because the metallic substrate of the disc is reflective, and the unburned dye prevents reflection while the holes in the dye permit it, allowing digital data to be represented. Concerns with digital audio recordingWord SizeThe number of bits used to represent a single audio wave (the word size) directly affects the distortion of a signal. Increasing a sample's word length by one bit doubles its possible values, likewise increasing the potential accuracy of each sample and the fidelity of the recording to the original. 24-bit recording is generally considered a current practical limit as this word length allows a signal-to-noise ratio exceeding that of most analog circuitry, which by necessity must be used in at least two points in the recording/playback chain. Sample rateThe sample rate is even more important a consideration than the word size. If the sample rate is too low, the sampled signal cannot be reconstructed to the original sound signal. Hence the output will be different from the input. The process of under sampling results in aliasing whereby the high frequency components of the sound wave are represented as being lower than they should be. This causes the output wave shape to be severely altered. To overcome aliasing, the sound signal (or other signal) must be sampled at a rate at least twice that of the highest frequency component in the signal. This is known as the Nyquist-Shannon sampling theorem. Error RectificationOne of the advantages of digital recording over analog recording is its resistance to errors. See also
|
| All Right Reserved © 2007, Designed by Stylish Blog. |