Digital audio systems may include compression, storage, processing, and transmission components. Conversion to a digital format allows convenient manipulation, storage, transmission, and retrieval of an audio signal. Unlike analog audio, in which making copies of a recording results in generation loss and degradation of signal quality, digital audio allows an infinite number of copies to be made without any degradation of signal quality.
Overview
A sound wave, in red, represented digitally, in blue (after sampling and 4-bit quantization).
Digital audio technologies are used in the recording, manipulation, mass-production, and distribution of sound, including recordings of songs, instrumental pieces, podcasts, sound effects, and other sounds. Modern online music distribution depends on digital recording and data compression. 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.[1] Before digital audio, the music industry distributed and sold music by selling physical copies in the form of records and cassette tapes. With digital audio and online distribution systems such as iTunes, companies sell digital sound files to consumers, which the consumer receives over the Internet. Popular streaming services such as Apple Music, Spotify, or YouTube, offer temporary access to the digital file, and are now the most common form of music consumption.[2]
An analog audio system converts physical waveforms of sound into electrical representations of those waveforms by use of a transducer, such as a microphone. The sounds are then stored on an analog medium such as magnetic tape, or transmitted through an analog medium such as a telephone line or radio. 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. Disturbances in a digital system do not result in error unless they are so large as to result in a symbol being misinterpreted as another symbol or disturbing the sequence of symbols. It is, therefore, generally possible to have an entirely error-free digital audio system in which no noise or distortion is introduced between conversion to digital format and conversion back to analog.[a]
A digital audio signal may be encoded for correction of any errors that might occur in the storage or transmission of the signal. This technique, known as channel coding, is essential for broadcast or recorded digital systems to maintain bit accuracy. Eight-to-fourteen modulation is the channel code used for the audio compact disc (CD).
Conversion process
The lifecycle of sound from its source, through an ADC, digital processing, a DAC, and finally as sound again.
If an audio signal is analog, a digital audio system starts with an ADC that converts an analog signal to a digital signal.[b] The ADC runs at a specified sampling rate and converts at a known bit resolution. CD audio, for example, has a sampling rate of 44.1kHz (44,100samples per second), and has 16-bit resolution for each stereo channel. Analog signals that have not already been bandlimited must be passed through an anti-aliasing filter before conversion, to prevent the aliasing distortion that is caused by audio signals with frequencies higher than the Nyquist frequency (half the sampling rate).
For playback, digital audio must be converted back to an analog signal with a DAC. According to the Nyquist–Shannon sampling theorem, with some practical and theoretical restrictions, a band-limited version of the original analog signal can be accurately reconstructed from the digital signal.
During conversion, audio data can be embedded with a digital watermark to prevent piracy and unauthorized use. Watermarking is done using a direct-sequence spread-spectrum (DSSS) method. 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. The strength of the embedding determines the strength of the watermark on the audio data.[4]
PCM was used in telecommunications applications long before its first use in commercial broadcast and recording. Commercial digital recording was pioneered in Japan by NHK and Nippon Columbia and their Denon brand, in the 1960s. The first commercial digital recordings were released in 1971.[16]
The BBC also began to experiment with digital audio 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]
The first 16-bit PCM recording in the United States was made by Thomas Stockham at the Santa Fe Opera in 1976, on a Soundstream recorder. An improved version of the Soundstream system was used to produce several classical recordings by Telarc in 1978. The 3M digital multitrack recorder in development at the time was based on BBC technology. The first all-digital album recorded on this machine was Ry Cooder's Bop till You Drop in 1979. British record label Decca began development of its own 2-track digital audio recorders in 1978 and released the first European digital recording in 1979.[16]
Popular professional digital multitrack recorders produced by Sony/Studer (DASH) and Mitsubishi (ProDigi) in the early 1980s helped to bring about digital recording's acceptance by the major record companies. Machines for these formats had their own transports built-in as well, using reel-to-reel tape in either 1/4", 1/2", or 1" widths, with the audio data being recorded to the tape using a multi-track stationary tape head. PCM adaptors allowed for stereo digital audio recording on a conventional NTSC or PAL video tape recorder.
The 1982 introduction of the CD by Philips and Sony popularized digital audio with consumers.[16]
ADAT became available in the early 1990s, which allowed eight-track 44.1 or 48kHz recording on S-VHS cassettes, and DTRS performed a similar function with Hi8 tapes.
Formats like ProDigi and DASH were referred to as SDAT (stationary-head digital audio tape) formats, as opposed to formats like the PCM adaptor-based systems and Digital Audio Tape (DAT), which were referred to as RDAT (rotating-head digital audio tape) formats, due to their helical-scan process of recording.
Like the DAT cassette, ProDigi and DASH machines also accommodated the obligatory 44.1kHz sampling rate, but also 48kHz on all machines, and eventually a 96kHz sampling rate. 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.
Unlike analog systems, modern digital audio workstations and audio interfaces allow as many channels in as many different sampling rates as the computer can effectively run at a single time. Avid Audio and Steinberg released the first digital audio workstation software programs in 1989.[17] Digital audio workstations make multitrack recording and mixing much easier for large projects which would otherwise be difficult with analog equipment.
Digital audio in recording applications is stored on audio-specific technologies including CD, DAT, Digital Compact Cassette (DCC) and MiniDisc. Digital audio may be stored in a standard audio file formats and stored on a Hard disk recorder, Blu-ray or DVD-Audio. Files may be played back on smartphones, computers or MP3 player. Digital audio resolution is measured in audio bit depth. Most digital audio formats use either 16-bit, 24-bit, and 32-bit resolution.
USB and IEEE 1394 (FireWire) for Real-Time Digital Audio Original Content: Mentions USB interfaces' popularity due to their small size and ease of use, and IEEE 1394 for digital audio. Relevance Check: This is directly relevant to the topic, as USB and FireWire are key interfaces for real-time digital audio in both consumer and professional audio applications. Action: Keep this section as is. 2. Audio Over Ethernet and Professional Protocols Original Content: Mentions various audio-over-Ethernet protocols and audio over IP in broadcasting and telephony. Relevance Check: Relevant to professional audio environments where Ethernet and IP-based audio protocols are commonly used. This covers systems for both broadcast (audio over IP) and telephony (VoIP) audio. Action: Keep the reference to audio-over-Ethernet and audio-over-IP technologies as they are highly relevant in professional contexts. 3. TDIF (TASCAM Proprietary Format) Original Content: Includes TDIF, a TASCAM format, using D-sub cables. Relevance Check: This is a highly specific and relevant mention in professional audio, especially for multi-channel setups where TDIF is still used in some high-end audio systems. Action: Retain this information. 4. Mic and Speaker Issues (Fix My Mic Speaker) Original Content: Discusses the issue of muffled sounds due to dust or water, and how the "Fix My Mic Speaker" tool helps clean the speaker and remove water. Relevance Check: This section appears a bit disconnected from the rest of the technical content on digital audio interfaces. It seems more focused on consumer device troubleshooting (like phones or laptops) rather than professional audio equipment. Action: The section is relevant to audio issues but less so in the context of professional audio interfaces. If the focus is on professional gear (as indicated by the rest of the content), this part might be better placed separately or omitted unless you're making a broader point about device maintenance. 5. Digital Audio-Specific Interfaces Original Content: Lists various digital audio interfaces such as A2DP, AC'97, ADAT, AES3, etc. Relevance Check: This section is highly relevant to the overall discussion. Each of these interfaces is integral to various audio applications, both in consumer and professional settings. It covers a broad range of interface types, from Bluetooth streaming (A2DP) to multi-channel professional standards (AES3, MADI, S/PDIF). Action: This section fits well and should remain intact, though it could be slightly streamlined to avoid redundancy. Suggestions for Greater Relevance and Flow: Mic and Speaker Troubleshooting: Since the focus of the article is primarily on audio interfaces and professional audio technologies, the section on "Fix My Mic Speaker" could be adjusted to make it relevant to professional audio gear. If you want to maintain it in the article, consider rephrasing it as a brief mention of how device maintenance (e.g., cleaning connectors or ensuring water/moisture protection) is important for the longevity and quality of professional audio interfaces and microphones.
Contextual Linking: To maintain a coherent flow, consider linking the mic and speaker troubleshooting section back to the broader theme of maintaining audio equipment for better sound quality, ensuring all parts of the system (hardware and software) are in optimal condition.
Revised Text with Adjusted Relevance: Digital Audio Interfaces: USB, IEEE 1394, and Other Protocols USB and IEEE 1394 (FireWire) have become essential for real-time digital audio in personal computing. USB interfaces are especially popular among independent audio engineers and producers due to their compact form, versatility, and ease of use. These interfaces are found in consumer audio equipment and support audio transfer based on AES3 standards.
For more professional setups, particularly in architectural and installation applications, several audio-over-Ethernet protocols provide high-quality, reliable transmission of audio over networks. These technologies are standard in the broadcasting sector, where audio over IP is favored for transmitting digital audio across various devices and platforms. Additionally, Voice over IP (VoIP) is widely used in telephony to deliver digital voice communications with high audio fidelity.
Specialized formats like TDIF (TASCAM's proprietary format using D-sub cables) are also used in multi-channel professional audio environments, allowing for robust, high-fidelity audio connections.
Ensuring Optimal Sound Quality: Mic and Speaker Maintenance Clear audio from your device’s microphone and speakers is essential for quality calls and sound production. In both consumer and professional audio systems, common issues such as dust accumulation or moisture in the speakers can cause muffled or distorted sound. If your microphone or speakers are not producing clear sound, it’s important to regularly clean and maintain the hardware. Tools designed to remove dust and moisture, such as Fix My Mic Speaker, can improve the sound quality by clearing blockages and ejecting water from the speaker area.
Whether working with professional audio gear or consumer devices, ensuring that the microphone and speaker areas are free from obstruction is crucial for preserving sound quality. Dust or water can dampen performance, affecting both hardware longevity and audio clarity.
Digital-Audio Specific Interfaces In addition to USB and FireWire, several other digital audio interfaces are commonly used across both consumer electronics and professional settings:
A2DP via Bluetooth, for high-quality audio streaming to wireless devices. AC'97, a legacy interface found on older PC motherboards, offering basic audio features. ADAT Lightpipe, a fiber-optic interface for multi-channel digital audio. AES3, an industry-standard professional audio interface using XLR connectors. AES47, a professional extension of AES3, designed to transmit digital audio over ATM networks. Intel High Definition Audio, a modern replacement for AC'97, supporting more channels and higher fidelity. I²S, used for inter-chip audio communication in consumer electronics. MADI, a robust interface for multi-channel digital audio in professional environments. MIDI, used for transmitting digital instrument data (not audio, but relevant for musicproduction). S/PDIF, commonly used for transmitting high-quality audio over coaxial or fiber-optic connections. These interfaces, ranging from legacy standards like AC'97 to modern technologies like AES3 and S/PDIF, are foundational for delivering high-quality audio in both consumer electronics and professional environments such as studios, live sound, and broadcast.
Final Verdict: Relevance: The technical sections on USB, IEEE 1394, and the various professional audio protocols are fully relevant to the topic of digital audio interfaces. The mention of mic [20] issues could use a slight contextual adjustment to better tie it into the theme of professional audio equipment maintenance. Flow: The revised version integrates all information in a way that maintains both technical accuracy and a cohesive narrative, making the article relevant for an audience interested in digital audio interfaces, while not deviating into overly consumer-centric details.
↑ Anti-alias filtering and optional digital signal processing may degrade the audio signal via passband ripple, non-linear phase shift, numeric precision quantization noise or time distortion of transients. However, these potential degradations can be limited by careful digital design.[3]
↑ Some audio signals such as those created by digital synthesis originate entirely in the digital domain, in which case analog to digital conversion does not take place.
Related Research Articles
A sound card is an internal expansion card that provides input and output of audio signals to and from a computer under the control of computer programs. The term sound card is also applied to external audio interfaces used for professional audio applications.
Digital Audio Tape is a signal recording and playback medium developed by Sony and introduced in 1987. In appearance it is similar to a Compact Cassette, using 3.81 mm / 0.15" magnetic tape enclosed in a protective shell, but is roughly half the size at 73 mm × 54 mm × 10.5 mm. The recording is digital rather than analog. DAT can record at sampling rates equal to, as well as higher and lower than a CD at 16 bits quantization. If a comparable digital source is copied without returning to the analogue domain, then the DAT will produce an exact clone, unlike other digital media such as Digital Compact Cassette or non-Hi-MD MiniDisc, both of which use a lossy data-reduction system.
S/PDIF is a type of digital audio interface used in consumer audio equipment to output audio over relatively short distances. The signal is transmitted over either a coaxial cable using RCA or BNC connectors, or a fibre-optic cable using TOSLINK connectors. S/PDIF interconnects components in home theaters and other digital high-fidelity systems.
Sound can be recorded and stored and played using either digital or analog techniques. Both techniques introduce errors and distortions in the sound, and these methods can be systematically compared. Musicians and listeners have argued over the superiority of digital versus analog sound recordings. Arguments for analog systems include the absence of fundamental error mechanisms which are present in digital audio systems, including aliasing and associated anti-aliasing filter implementation, jitter and quantization noise. Advocates of digital point to the high levels of performance possible with digital audio, including excellent linearity in the audible band and low levels of noise and distortion.
AES3 is a standard for the exchange of digital audio signals between professional audio devices. An AES3 signal can carry two channels of pulse-code-modulated digital audio over several transmission media including balanced lines, unbalanced lines, and optical fiber.
Sound quality is typically an assessment of the accuracy, fidelity, or intelligibility of audio output from an electronic device. Quality can be measured objectively, such as when tools are used to gauge the accuracy with which the device reproduces an original sound; or it can be measured subjectively, such as when human listeners respond to the sound or gauge its perceived similarity to another sound.
Direct Stream Digital (DSD) is a trademark used by Sony and Philips for their system for digitally encoding audio signals for the Super Audio CD (SACD).
In digital recording, an audio or video signal is converted into a stream of discrete numbers representing the changes over time in air pressure for audio, or chroma and luminance values for video. This number stream is saved to a storage device. To play back a digital recording, the numbers are retrieved and converted back into their original analog audio or video forms so that they can be heard or seen.
The digital sound revolution refers to the widespread adoption of digital audio technology in the computer industry beginning in the 1980s.
The 8mm video format refers informally to three related videocassette formats. These are the original Video8 format and its improved successor Hi8, as well as a more recent digital recording format known as Digital8. Their user base consisted mainly of amateur camcorder users, although they also saw important use in the professional television production field.
The Digital Audio Stationary Head or DASH standard is a reel-to-reel, digital audio tape format introduced by Sony in early 1982 for high-quality multitrack studio recording and mastering, as an alternative to analog recording methods. DASH is capable of recording two channels of audio on a quarter-inch tape, and 24 or 48 tracks on 1⁄2-inch-wide (13 mm) tape on open reels of up to 14 inches. The data is recorded on the tape linearly, with a stationary recording head, as opposed to the DAT format, where data is recorded helically with a rotating head, in the same manner as a VCR. The audio data is encoded as linear PCM and boasts strong cyclic redundancy check (CRC) error correction, allowing the tape to be physically edited with a razor blade as analog tape would, e.g. by cutting and splicing, and played back with no loss of signal. In a two-track DASH recorder, the digital data is recorded onto the tape across nine data tracks: eight for the digital audio data and one for the CRC data; there is also provision for two linear analog cue tracks and one additional linear analog track dedicated to recording time code.
A PCM adaptor is a device that encodes digital audio as video for recording on a videocassette recorder. The adapter also has the ability to decode a video signal back to digital audio for playback. This digital audio system was used for mastering early compact discs.
Soundstream Inc. was the first United States audiophile digital audio recording company, providing commercial services for recording and computer-based editing.
The history of sound recording - which has progressed in waves, driven by the invention and commercial introduction of new technologies — can be roughly divided into four main periods:
The dbx Model 700 Digital Audio Processor was a professional audio ADC/DAC combination unit, which digitized a stereo analog audio input into a bitstream, which was then encoded and encapsulated in an analog composite video signal, for recording to tape using a VCR as a transport. Unlike other similar pieces of equipment like the Sony PCM-F1, the Model 700 used a technique called Companded Predictive Delta Modulation, rather than the now-common pulse-code modulation. At the time of its introduction in the mid-1980s the device was the first commercial product to use this method, although it had been proposed in the 1960s and prototyped in the late '70s.
In digital audio using pulse-code modulation (PCM), bit depth is the number of bits of information in each sample, and it directly corresponds to the resolution of each sample. Examples of bit depth include Compact Disc Digital Audio, which uses 16 bits per sample, and DVD-Audio and Blu-ray Disc, which can support up to 24 bits per sample.
An audio signal is a representation of sound, typically using either a changing level of electrical voltage for analog signals, or a series of binary numbers for digital signals. Audio signals have frequencies in the audio frequency range of roughly 20 to 20,000 Hz, which corresponds to the lower and upper limits of human hearing. Audio signals may be synthesized directly, or may originate at a transducer such as a microphone, musical instrument pickup, phonograph cartridge, or tape head. Loudspeakers or headphones convert an electrical audio signal back into sound.
Adaptive differential pulse-code modulation (ADPCM) is a variant of differential pulse-code modulation (DPCM) that varies the size of the quantization step, to allow further reduction of the required data bandwidth for a given signal-to-noise ratio.
Audio connectors and video connectors are electrical or optical connectors for carrying audio or video signals. Audio interfaces or video interfaces define physical parameters and interpretation of signals. For digital audio and digital video, this can be thought of as defining the physical layer, data link layer, and most or all of the application layer. For analog audio and analog video these functions are all represented in a single signal specification like NTSC or the direct speaker-driving signal of analog audio.
Pulse-code modulation (PCM) is a method used to digitally represent analog signals. It is the standard form of digital audio in computers, compact discs, digital telephony and other digital audio applications. In a PCM stream, the amplitude of the analog signal is sampled at uniform intervals, and each sample is quantized to the nearest value within a range of digital steps. Alec Reeves, Claude Shannon, Barney Oliver and John R. Pierce are credited with its invention.
↑ USpatent 2605361,C. Chapin Cutler,"Differential Quantization of Communication Signals",issued 1952-07-29
↑ P. Cummiskey, Nikil S. Jayant, and J. L. Flanagan, "Adaptive quantization in differential PCM coding of speech", Bell Syst. Tech. J., vol. 52, pp. 1105—1118, Sept. 1973
↑ Cummiskey, P.; Jayant, Nikil S.; Flanagan, J. L. (1973). "Adaptive quantization in differential PCM coding of speech". The Bell System Technical Journal. 52 (7): 1105–1118. doi:10.1002/j.1538-7305.1973.tb02007.x. ISSN0005-8580.
1 2 3 Schroeder, Manfred R. (2014). "Bell Laboratories". Acoustics, Information, and Communication: Memorial Volume in Honor of Manfred R. Schroeder. Springer. p.388. ISBN9783319056609.
↑ J. P. Princen, A. W. Johnson und A. B. Bradley: Subband/transform coding using filter bank designs based on time domain aliasing cancellation, IEEE Proc. Intl. Conference on Acoustics, Speech, and Signal Processing (ICASSP), 2161–2164, 1987.
This page is based on this Wikipedia article Text is available under the CC BY-SA 4.0 license; additional terms may apply. Images, videos and audio are available under their respective licenses.
