Quote:
Originally Posted by AleksandarK
Da ne otvaram novi. Ukratko, kako povecati dinamicki opseg digitalnim hardverskim sintisajzerima (rompleri, VA) pri usnimavanju (ili uzivo) ?
Zvuk digitalne tehnike grubo receno vidim kao 2D sliku analogno-elektromehanickog zvuka. Tanak i komprimovan (ovo je grubo receno da bih istakao ideju). Kako npr. zvuk DX e.piano procesirati da u snimku bude velicine pravog Rhodesa ili Wurlitzera ?
Probao bih provlacenje kroz lampaske preampove, EQ, FX, kompresor. Ima li ko da podeli svoja iskustva sa konkretnim spravama, da ne lutam previse ?
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Druze, malo si pobrkao lonchice. Evo procitaj za pocetak ovo, pa nastavi dalje..
Dynamic range
Dynamic range is the difference between the largest and smallest signal a system can record or reproduce. With the proper application of
dither, digital systems can reproduce signals with levels lower than their resolution would normally allow
Digital audio
A set of digital audio samples contains data that, when converted into an
analog signal, provides the necessary information to reproduce the
sound wave. In
pulse-code modulation (PCM) sampling, the bit depth will limit
signal-to-noise ratio (S/N). The bit depth will not limit
frequency range, which is limited by the
sample rate.
By increasing the sampling bit depth,
quantization noise is reduced so that the S/N is improved. The '
rule-of-thumb' relationship between bit depth and S/N is, for each 1-bit increase in bit depth, the S/N will increase by 6
dB.
[2][3] 24-bit digital audio has a theoretical maximum S/N of 144 dB, compared to 96 dB for 16-bit; however, as of 2007 digital audio converter technology is limited to a S/N of about 124 dB (21-bit)
[4] because of real world limitations in
integrated circuit design. Still, this approximately matches the performance of the
human ear.
[5][6]
Technically speaking, bit depth is only meaningful when applied to pure PCM devices. Non-PCM formats, such as
lossy compression systems like
MP3, have bit depths that are not defined in the same sense as PCM. In lossy audio compression, where bits are allocated to other types of information, the bits actually allocated to individual samples are allowed to fluctuate within the constraints imposed by the allocation algorithm.
24-bit quantization
24-bit audio is sometimes used undithered, because for most audio equipment and situations the noise level of the digital converter can be louder than the required level of any dither that might be applied.
There is some disagreement over the recent trend towards higher bit-depth audio. It is argued by some that the dynamic range presented by 16-bit is sufficient to store the dynamic range present in almost all music. In terms of pure data storage this is often true, as a high-end system can extract an extremely good sound out of the 16-bits stored in a
well-mastered CD. However, audio with very loud and very quiet sections can require some of the above dithering techniques to fit it into 16-bits. This is not a problem for most recently produced popular music, which is often mastered so that it constantly sits close to the maximum signal (see
loudness war); however, higher resolution audio formats are already being used (especially for applications such as film soundtracks, where there is often a very wide dynamic range between whispered conversations and explosions).
For most situations the advantage given by resolution higher than 16-bit is mainly in the processing of audio. No digital filter is perfect, but if the audio is upsampled and the audio is done in 24-bit or higher, then the distortion introduced by filtering will be much quieter (as the errors always creep into the
least significant bits) and a well-designed filter can weight the distortion more towards the higher inaudible frequencies (but a sample rate higher than 48kHz is needed so that these inaudible ultrasonic frequencies are available for soaking up errors).
There is also a good case for 24-bit (or higher) recording in the live studio, because it enables greater headroom (often 24dB or more rather than 18dB) to be left on the recording without encountering quantization errors at low volumes. This means that brief peaks are not harshly clipped, but can be compressed or soft-limited later to suit the final medium.
Environments where large amounts of signal processing are required (such as mastering or synthesis) can require even more than 24 bits. Some modern audio editors convert incoming audio to 32-bit (both for an increased dynamic range to reduce clipping, and to minimize noise in intermediate stages of filtering).