Digital audio

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Digital audio is the term used for digitized audio signals (music, etc.).

Analog-to-digital conversion

Sounds are sound waves . You can use a sound transducer such as B. a microphone are converted into analog electrical signals to be transmitted, processed or recorded.

Digitization means transforming these analog signals into discrete values. For this purpose, the values ​​of the audio signals must be sampled and stored sufficiently often. According to the sampling theorem known in digital signal processing , the sampling frequency must be more than twice as large as the highest frequency of the signal to be sampled. If scanning is too seldom, the so-called aliasing effect occurs . Since humans only hear signals up to approx. 20 kHz, audio signals for CDs are sampled at 44.1 kHz. Higher frequencies in the audio signal must be filtered (analog) beforehand, otherwise errors will occur. The aliasing effect would "mirror" the frequencies that are too high to the lower frequencies.

DVDs and DAT work at 48 kHz, newer formats and professional music productions even work at up to 192 kHz.

High sampling rates not only increase the maximum reproducible frequency, but also facilitate the digital processing of the audio material: if the available sample rate is higher than the required one, digital audio effects can use more samples for the calculation and thus e.g. B. Interpolate speed stretches better. More on this under oversampling . However, a disadvantage in production is the disproportionately increasing load on the DAW due to the large amounts of data. Depending on the hardware used, larger buffers must be selected so that the latency increases.

Coding

Large amounts of data are generated for uncompressed, PCM-encoded audio data. The required storage space per second results from:

  • Bits per second = sample rate × sample width × channels

For a CD (44.1 kHz, 16 bit, 2 channels) this results in 1411200 bits per second or 10.1  MiB / min.

44100 Hz × 16 bits × 2 = 1411200 bits per second
÷ 8 = 176400 bytes per second
÷ 1024 ≈ 172.2656  KiB / s
÷ 1024 ≈ 0.1682 MiB / s
× 60 ≈ 10.0936 MiB / min

To reduce this amount of data, formats such as Ogg / Vorbis or MP3 were created, which reduce the amount of data through lossy processing. Depending on the desired output size, the quality fluctuates between not noticeable and severely alienated ; One speaks of the compression artifact : if it is audible, the selected resulting bit rate is too low and the result can be clearly distinguished from the original.

Music exchanges

The emergence and private use of the Internet made it possible to spread digital audio files on a previously unseen scale. File sharing networks like Napster contained hundreds of thousands of music files. Often they are the only source of sound carriers that are no longer offered by record companies.

Playback devices

Audio players enable the playback of compressed audio files on the computer. But portable players have also been on the market for a number of years with constantly growing memory and functionality. Often, however, these devices can only handle MP3 or WMA . Since WMA is a proprietary data format and MP3 is patented, the manufacturing companies can restrict or prevent the use of audio files or add additional costs by changing the license policy. WMA has long been talking about pay-per-play , which means that each individual playback of the song is chargeable or music is only rented. Some devices therefore use the patent-free Ogg / Vorbis format, for which no license fees can be charged, as it is publicly available to everyone.

See also

literature

  • Thomas Görne: Sound engineering. 1st edition, Carl Hanser Verlag, Leipzig, 2006, ISBN 3-446-40198-9 .
  • Roland Enders: The home recording manual. 3rd edition, Carstensen Verlag, Munich, 2003, ISBN 3-910098-25-8 .

Web links