Digital audio
Conversion to a digital format allows convenient manipulation, storage, transmission, and retrieval of an audio signal.
The availability of music as data files, rather than as physical objects, has significantly reduced the costs of distribution as well as making it easier to share copies.
The process is reversed for reproduction: the electrical audio signal is amplified and then converted back into physical waveforms via a loudspeaker.
Analog audio retains its fundamental wave-like characteristics throughout its storage, transformation, duplication, and amplification.
Analog audio signals are susceptible to noise and distortion, due to the innate characteristics of electronic circuits and associated devices.
This technique, known as channel coding, is essential for broadcast or recorded digital systems to maintain bit accuracy.
The audio information is then modulated by a pseudo-noise (PN) sequence, then shaped within the frequency domain and put back in the original signal.
[5] In 1950, C. Chapin Cutler of Bell Labs filed the patent on differential pulse-code modulation (DPCM),[6] a data compression algorithm.
Adaptive DPCM (ADPCM) was introduced by P. Cummiskey, Nikil S. Jayant and James L. Flanagan at Bell Labs in 1973.
[9] Initial concepts for LPC date back to the work of Fumitada Itakura (Nagoya University) and Shuzo Saito (Nippon Telegraph and Telephone) in 1966.
[10] During the 1970s, Bishnu S. Atal and Manfred R. Schroeder at Bell Labs developed a form of LPC called adaptive predictive coding (APC), a perceptual coding algorithm that exploited the masking properties of the human ear, followed in the early 1980s with the code-excited linear prediction (CELP) algorithm.
Commercial digital recording was pioneered in Japan by NHK and Nippon Columbia and their Denon brand, in the 1960s.
By the early 1970s, it had developed a 2-channel recorder, and in 1972 it deployed a digital audio transmission system that linked their broadcast center to their remote transmitters.
[16] ADAT became available in the early 1990s, which allowed eight-track 44.1 or 48 kHz recording on S-VHS cassettes, and DTRS performed a similar function with Hi8 tapes.
They overcame the problems that made typical analog recorders unable to meet the bandwidth (frequency range) demands of digital recording by a combination of higher tape speeds, narrower head gaps used in combination with metal-formulation tapes, and the spreading of data across multiple parallel tracks.
[17] Digital audio workstations make multitrack recording and mixing much easier for large projects which would otherwise be difficult with analog equipment.
[18][19] The silicon-gate CMOS (complementary MOS) PCM codec-filter chip, developed by David A. Hodges and W.C. Black in 1980,[18] has since been the industry standard for digital telephony.
