Digital Audio Coding
Digital audio technologies are used to record, process, mass produce, and distribute sound, including the recording of songs, instrumental pieces, podcasts, sound effects, and other sounds.
Today’s online music distribution relies on digital recording and data compression. The availability of music as data files instead of physical objects has significantly reduced distribution costs. Before the advent of digital sound, the music industry distributed and sold music, selling physical copies in the form of records and cassettes.
Using online and digital audio distribution systems such as iTunes, companies sell digital audio files to consumers that the consumer receives over the Internet. An analog audio system converts the physical waveforms of sound into electrical representations of these waveforms using a transducer such as a microphone. The sounds are then stored on analog media, such as magnetic tape, or transmitted through analog media, such as a telephone line or radio. For playback, the process is reversed: an electrical audio signal is amplified and then converted back to physical waveforms through a speaker.
Analog audio retains its fundamental waveform characteristics when stored, converted, dubbed, and amplified. Analog audio signals are prone to noise and distortion due to the inherent characteristics of electronic circuits and related devices. Interference in a digital system does not result in an error, unless the interference is large enough to cause one character to be misinterpreted as another character or to be out of sequence.
Therefore, it is generally possible to have a completely error-free digital audio system in which there is no noise or distortion between converting to digital and converting to analog. The digital audio signal can be further encoded to correct any errors that may occur during storage or transmission of the signal. This technique, known as channel coding, is necessary for broadcast or recorded digital systems to maintain bit fidelity. Modulation from eight to fourteen is a channel code used on audio CDs. Conversion process
The life cycle of sound from its source through ADC, digital processing, DAC, and finally again as sound. A digital audio system begins with an ADC, which converts an analog signal into digital. The ADC operates at the specified sample rate and converts to a known bit resolution.
For example, CD audio has a sample rate of 44.1 kHz (44,100 samples per second) and a resolution of 16 bits for each stereo channel. Analog signals that have not yet been band-limited should be damped before conversion to avoid interpolation distortion caused by audio signals above the Nyquist frequency (half the sample rate).
The digital audio signal can be stored or transmitted. Digital audio can be stored on a CD, digital audio player, hard drive, USB flash drive, or any other digital storage device. The digital signal can be modified by digital signal processing, where effects can be filtered or applied. Frequency transform Sample rates, including increasing and decreasing the sample rate, can be used to match signals that have been encoded with a different sample rate to a common preprocessing sample rate. Audio compression methods such as MP3, Advanced Audio Coding, Ogg Vorbis, or FLAC are commonly used to reduce file size.
Digital audio can be transmitted through digital audio interfaces such as AES3 or MADI. Digital audio can be transmitted over a network using Audio over Ethernet, Audio over IP, or other standards and media transmission systems. For playback, digital audio must be converted back to an analog signal using a DAC. According to the Nyquist-Shannon Sampling Theorem, with some practical and theoretical limitations, the bandwidth-limited version of the original analog signal can be accurately reconstructed from the digital signal. –
