Beruflich Dokumente
Kultur Dokumente
A to D
2 - multiple 1 (mic) 2 (L & R) 2 (L & R) 2 - multiple 2 (L & R) 2 (L & R) 2 (L & R) Multiple 8 2 (L & R)
D to A (DAC)
2 - multiple 2 (L & R headphones) Multiple (individual & stereo?) 2 (L & R) 2- multiple 2 (L & R) 2 (L & R) 2 (L & R) 2 (L & R) Multiple 8 2 (L & R) 2 (L & R) 2 (L & R) 2 (L & R)
Image quality
All image designers know that quality of an image depends on 2 factors also... Number of pixels per inch (ppi resolution) Number of colours in the images palette (determined by bit depth) Creating a good quality digital audio signal depends on 2 similar parameters.
1. What is a sample?
Although it is common to use the word "sample" to refer to a complete sound (perhaps a piano note or drum break/loop), in digital theory: a "sample" is a single measurement of amplitude. A sample may also be referred to as a... Snapshot Sample measurement
Format
Audio CD DVD Professional multi-track recording (Logic Pro, ProTools) MP3s
Sample rate
44,100 samples per second (44.1KHz) Up to 96,000 (96KHz) 48KHz or 96KHz, and even sometimes 192KHz Variety of sample rates. The trade-off is always between quality and file size.
Nyquist theory
During his research into digital audio in the first half of the 20th century, Harry Nyquist (a scientist) produced a simple rule that should be followed to determine appropriate sample rates for differing sounds. "The sample rate should be a little over twice the amount of the highest audio frequency (harmonic) to be recorded if poor sound quality is to be avoided". Because humans can hear audio frequencies as high as 20KHz (20,000cps/Hz), a minimum sample rate of 44.1KHz (or 44,100 sample measurements a second) was decided upon... Human audio spectrum = 20Hz to 20,000Hz (20KHz) ... therefore... Highest audio frequency = 20,000Hz ... therefore... 20,000 x 2 = 40,000 + "a little bit more" = 44,100 samples per second At the time, 44.1KHz was considered the best compromise of quality and file size. Over the last 20 years, there has been much debate among audio engineers and designers over the importance of using higher samples rates. Although many can't hear the difference between 44.1 and 192KHz audio (myself included!), others claim they can and that because equipment can easily handle higher rates, why not use them? Given the quality of much modern domestic audio equipment (iPods, car stereos, phones, digital TV, DAB radio etc) produces increasingly inferior sound, it is unlikely that many consumers will benefit anyway.
Aliasing
If the sample rate is set too low (ie less than 2 times the highest audio frequency to be recorded), a type of distortion called "aliasing" will be audible in the signal when it is converted back to analogue by a DAC (digital to analogue converter). Consider a soundwave/harmonic at a low frequency of 20Hz. There will be 20 cycles of its waveform every second. This means that if it is recorded at a sample rate of 44.1KHz, each cycle will be represented by 2,205 samples. 44,100 divided by 20 = 2,205. So each cycle of a low frequency soundwave/harmonic is measured comprehensively and the shape of the waveform is recorded accurately (Diagram 1). Diagram 1 - accurate sampling: Sampling (A to D) at 44.1KHz of one 20Hz cycle (low bass)
Diagram 3 - adequate playback: Playback (A to D) at 44.1KHz of one 20,000Hz cycle (hi treble)
Aliasing noise
There is an effect in film making caused by its fixed frame rate (24 frames per second) that can lead to the odd visible effect of a speeding cars wheels appearing to revolve backwards. This happens because 24fps is insufficient to capture fast motion. This is called (visual) aliasing. In digital audio recording, if the recording sample rate is set lower than the required minimum 44.1kHz (for the high frequency soundwave/harmonics), then the soundwave produced by the D to A conversion at playback will be disastrously different. The wave is changed to a lower frequency wave. In a complex soundwave containing many harmonics, only the harmonics for which the sample rate is insufficient will be altered. Harmonics for which the sample rate is adequate will reproduced accurately. The audible effect of this can be audible random noise or unpleasant and unwanted lower harmonics within the sound. The unwanted harmonics are known as Aliasing Noise. Consider a soundwave at 20,000Hz being recorded at a sample rate of 32kHz. This would mean 1.5 samples per cycle of the waveform, clearly inadequate (Diagram 4). Diagram 4 - inadequate sampling: Sampling (A to D) at 32KHz of two 20,000Hz cycles (hi treble)
Now look at the soundwave reconstructed by the D to A converter (Diagram 5). The wave shape has changed dramatically, the wave is a lower frequency, and the sound has been distorted. Diagram 5 - inadequate playback: Playback (D to A) at 32KHz of two 20,000Hz cycles (hi treble)
Anti-aliasing filters
Analogue to digital converters therefore employ a low pass filter before the converters to remove any harmonics from the soundwave which are above the highest frequency that the sample rate can accommodate. Thus, an anti-aliasing filter in a CD recorder will remove any harmonics above 20KHz from a soundwave before it is converted and recorded.
Jitter
Jitter refers to irregularities in the time intervals between samples. Jitter can occur when... the clock regulating the A to D conversion is not regular, this is the worse case scenario because jitter is "written" into the data stream a cable with (relatively) high capacitance, in which the samples (pulse wave) are traveling, adversely effects the wave shape the clock regulating the D to A conversion is not regular Therefore the accuracy of the digital clock, which governs when samples occur, is paramount. If the clock is not accurate, jitter will occur and the audio quality will suffer.
Bit depth
In audio files, higher bit depth means better sound quality. In short, higher bit depths provide a converter with a more accurate "ruler" (higher bit resolution) to measure amplitude with, thereby producing more accurate measurements. In audio quality terms, more accurate measurements mean less distortion of the true shape of the soundwave.
Quantisation
A "ruler" with 256 divisions, or "points of resolution", is NOT very accurate. If when a measurement is taken, the amplitude of the wave does not fall exactly on one of these points, then the measurement must be rounded up or down to the next nearest point. This process is called Quantisation and results in a distorted recording of the true shape of the wave.
Audio dithering
The process of converting a high bit depth audio signal to a lower one is most commonly referred to as truncating. Essentially some of the bits in each byte/sample are thrown away (the least significant bits to be precise).
The effect of this process on an audio signal is to "magnify" quantisation errors which can result in audible distortion, especially in the quieter parts of an audio signal. Audio dithering is a process whereby low level white noise (random sound) is introduced into the signal to help randomise quantisation errors. The effect of this is to turn the audible effects of quantisation errors from unpleasant distortion into a the more acceptable analogue noise. Dithering is most commonly used at the CD mastering stage of music production, but dither can be used for other reasons too. The following are some of the processes that involve dithering ...
Digital interconnection
When passing a signal digitally between two devices, such as a DAW and a digital mixer, the signal may be converted and dithered if the bit-depths of the two systems don't match (the sample rate must match, otherwise a sample rate converter will need to be used).
Digital processors
Some effect processors allow you to set parameters for dither which will be automatically be introduced in the signal if it drops below a certain level.
Advantages
Analogue Once recorded, audio is stored/archived. 2" 24 track tape is a world wide standard. Warm and natural sound? Theoretically better audio bandwidth. Tried and tested format. Editing limitations discourage constant tinkering and changing of audio.
Disadvantages
Cheaper recorders suffer from distortion and tape noise/hiss. Tape is expensive and vulnerable to deterioration. Tape is becoming increasingly hard to source. Recorded need constant maintenance. Linear format - tape must be wound/rewound to the location of the recording to be heard. Editing of audio difficult if not impossible. Harder to synchronise. Copying deteriorates sound. Computers can crash. Data can become corrupted. Data must be archived when hard drives become full.
Digital
Better sound for cheaper equipment. Potential for very low distortion and noise/hiss. Variety of recorder options (tape,
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