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Audio levels display on a digital audio recorder (Zoom H4n)

In digital recording, audio signals picked up by a microphone or other transducer or video signals picked up by a camera or similar device are converted into a stream of discrete numbers, representing the changes over time in air pressure for audio, and chroma and luminance values for video, then recorded to a storage device. To play back a digital sound recording, the numbers are retrieved and converted back into their original analog waveforms so that they can be heard through a loudspeaker. To play back a digital video recording, the numbers are retrieved and converted back into their original analog waveforms so that they can be viewed on a video monitor, television or other display.

TimelineEdit

ProcessEdit

Recording

  1. The analog signal is transmitted from the input device to an analog-to-digital converter (ADC).
  2. The ADC converts this signal by repeatedly measuring the momentary level of the analog (audio) wave and then assigning a binary number with a given quantity of bits (word length) to each measuring point.
  3. The frequency at which the ADC measures the level of the analog wave is called the sample rate or sampling rate.
  4. A digital audio sample with a given word length represents the audio level at one moment.
  5. The longer the word length the more precise the representation of the original audio wave level.
  6. The higher the sampling rate the higher the upper audio frequency of the digitized audio signal.
  7. The ADC outputs a sequence of digital audio samples that make up a continuous stream of 0s and 1s.
  8. These binary numbers are stored on recording media such as a hard drive, optical drive or in solid state memory.

Playback

  1. The sequence of numbers is transmitted from storage into a digital-to-analog converter (DAC), which converts the numbers back to an analog signal by sticking together the level information stored in each digital sample, thus rebuilding the original analog wave form.
  2. This signal is amplified and transmitted to the loudspeakers or video screen.

Recording of bitsEdit

Even 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 mediaEdit

For 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 optical disc recording technologies such as 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 recordingEdit

Word sizeEdit

The number of bits used to represent a sampled audio wave (the word size) directly affects the resulting noise in a recording after intentionally added dither, or the distortion of an undithered signal.[56]

The number of possible voltage levels at the output is simply the number of levels that may be represented by the largest possible digital number (the number 2 raised to the power of the number of bits in each sample). There are no “in between” values allowed. If there are more bits in each sample the waveform is more accurately traced, because each additional bit doubles the number of possible values. The distortion is roughly the percentage that the least significant bit represents out of the average value. Distortion (as a percentage) in digital systems increases as signal levels decrease, which is the opposite of the behavior of analog systems.[57]

Sample rateEdit

The sample rate is just as important a consideration as the word size. If the sample rate is too low, the sampled signal cannot be reconstructed to the original sound signal.

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.

For recording music-quality audio the following PCM sampling rates are the most common: 44.1, 48, 88.2, 96, 176.4, and 192 kHz.

When making a recording, experienced audio recording and mastering engineers will normally do a master recording at a higher sampling rate (i.e. 88.2, 96, 176.4 or 192 kHz) and then do any editing or mixing at that same higher frequency. High resolution PCM recordings have been released on DVD-Audio (also known as DVD-A), DAD (Digital Audio Disc—which utilizes the stereo PCM audio tracks of a regular DVD), DualDisc (utilizing the DVD-Audio layer), or Blu-ray (Profile 3.0 is the Blu-ray audio standard, although as of mid-2009 it is unclear whether this will ever really be used as an audio-only format). In addition it is nowadays also possible and common to release a high resolution recording directly as either an uncompressed WAV or lossless compressed FLAC file[58] (usually at 24 bits) without down-converting it.

However, if a CD (the CD Red Book standard is 44.1 kHz 16 bit) is to be made from a recording, then doing the initial recording using a sampling rate of 44.1 kHz is obviously one approach. Another approach that is usually preferred is to use a higher sample rate and then downsample to the final format's sample rate. This is usually done as part of the mastering process. One advantage to the latter approach is that way a high resolution recording can be released, as well as a CD and/or lossy compressed file such as mp3—all from the same master recording.

Beginning in the 1980s, music that was recorded, mixed and mastered digitally was often labelled using the SPARS code to describe which processes were analog and which were digital.

Error rectificationEdit

One of the advantages of digital recording over analog recording is its resistance to errors.

See alsoEdit

ReferencesEdit

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