Digital Audio Basics: Sample Rate and Bit Depth

Although discussions of digital audio conversion have filled several books, a fundamental understanding of two terms is particularly important to correctly using your computer-based recording system: sample rate and bit depth.

The conversion process is complex, and there are multiple ways to accomplish it. But no worries: We’re just going to discuss sample rate and bit depth at a basic level, as applied to linear pulse-code modulation (PCM), one of the most common conversion technologies.

Sampling Basics

At the most basic level, computers operate one step at a time by turning a succession of switches on or off at very high speed. Since computers “think” in discrete steps, in order to convert analog audio signals to the digital domain, it’s necessary to describe the continuous analog waveform mathematically as a succession of discrete amplitude values.

In an analog-to-digital converter, this is accomplished by capturing, at a fixed rate, a rapid series of short “snapshots”—samples —of a specified size. Each audio sample contains data that provides the information necessary to accurately reproduce the original analog waveform. Things like dynamic range, frequency content, and so on are all contained within this datastream. The instantaneous amplitude level in each sample is given the value of the nearest measuring increment—a process called quantization. By reproducing these values and playing them back in the same order and at the same rate at which they were captured, a digital-to-analog converter produces a practically identical (in theory) copy of the original waveform.

The rate of capture and playback is called the sample rate. The sample size—more accurately, the number of bits used to describe each sample—is called the bit depth or word length. The number of bits transmitted per second is the bit rate. Let’s take a look at this as it applies to digital audio.

Digging A Bit Deeper

The on/off status of each switch in a computer is represented as 1 or 0, a system known as binary. Thus, a string of binary digits—bits —is used to describe anything a computer does, including manipulating and displaying text, images, and audio. Computers can manage entire strings of these bits at a time; a group of 8 bits is known as a byte; one or more bytes compose a digital word. Sixteen bits (two bytes) means that there are 16 digits in a word, each of them a 1 or 0; 24 bits (three bytes) means that there are 24 binary digits per word; and so on.

The number of bits in a word determines how precise the values are. Working with a higher bit depth is like measuring with a ruler that has finer increments: you get a more precise measurement. When the values are in finer increments, the converter doesn’t have to quantize as much to get to the nearest measuring increment.

FIG.1: If the bit depth is low (a), the signal will be inaccurately converted because it’s sampled in large increments. By increasing the bit depth (b), you get finer increments and a more accurate representation of the signal.

Thus, a higher bit depth enables the system to accurately record and reproduce more subtle fluctuations in the waveform (see Fig. 1). The higher the bit depth, the more data will be captured to more accurately re-create the sound. If the bit depth is too low, information will be lost, and the reproduced sample will be degraded. For perspective, each sample recorded at 16-bit resolution can contain any one of 65,536 unique values (216). With 24- bit resolution, you get 16,777,216 unique values (224)—a huge difference!

The most important practical effect of bit depth is that it determines the dynamic range of the signal. In theory, 24-bit digital audio has a maximum dynamic range of 144 dB, compared to 96 dB for 16-bit but today’s digital audio converter technology cannot come close to that upper limit. As of this writing, the 24-bit converters in StudioLive™ (including StudioLive AI-series) digital mixers and the FireStudio™ Mobile interface offer a dynamic range of 118 dB, which is close to the best dynamic range attainable with current technology.

The Going Rate

FIG. 2: Increasing the sample rate but not the bit depth (a) improves the accuracy of the representation because the converter is taking “snapshots” of the signal more frequently. However, increasing both the sample rate and the bit depth (b) produces much more accurate results.

As noted earlier, in the digital conversion process, the converters record and play samples at specified sample rates. The Nyquist-Shannon sampling theorem states that in order to accurately reconstruct a signal of a specified bandwidth (that is, a definable frequency range, such as 20 Hz to 20 kHz), the sampling frequency must be greater than twice the highest frequency of the signal being sampled. If lower sampling rates are used, the original signal’s information may not be completely recoverable from the sampled signal (see Fig. 2).

If the sampling frequency is too low, aliasing distortion can result. Aliasing is a major concern when using analog-to-digital conversion. Improper sampling of the analog signal will cause high-frequency components of the signal to be aliased with
genuine lower-frequency components. If this happens, the digital-to-analog conversion will create an incorrectly reconstructed signal.

In addition, higher sampling rates enable you to record very high frequencies above the normal range of human hearing. While inaudible by themselves, these ultrasonic frequencies can interact, creating intermodulation distortion (such as beating) that results in audible frequency content that many engineers believe to impart subtle psychoacoustic effects.

For a variety of reasons, then, many recording engineers rely on sampling rates of 88.2, 96, and even 192 kHz to ensure extremely accurate recordings that capture every detail.

Which rate you choose depends at least in part on the product you need to deliver. For example, audio CDs and MP3s are delivered at 44.1 kHz, so sampling at 88.2 kHz makes the converter’s calculations relatively simple. Digital broadcast uses 48 kHz, so a 96 kHz sampling rate is an obvious choice. That said, some engineers believe that today’s sample-rate conversion is good enough that it’s not necessary to choose a rate based on keeping the math simple. For these engineers, the higher rate is generally considered better.

The High-Resolution Frontier

Finally, one often encounters the term “high-resolution audio” but it is rarely defined. That’s because there is no agreed-upon definition. For many years, “resolution” referred to bit depth, but in recent years, the term has been used more broadly to refer to both sample rate and bit depth. And “high resolution,” in particular, is a relative term. When 8-bit audio was in common use, 16-bit was “high resolution.” Today, 24-bit, 96 kHz audio is considered “high resolution.” In the future, it might be 32-bit, 192 kHz and beyond.

*sources

http://www.presonus.com/news/articles/sample-rate-and-bit-depth
http://www.ItchyTastyRecords.com

Secret Options for Ableton Live

Secret Options for Ableton Live 

Options.txt file for Live
Live Versions: 1 - 9 Operating System: ALL
With the "Options.txt" file we offer a way to change some of Live's options for special circumstances. These options are mainly used for developing and internal testing. However, some of you may find them useful as well.


If you have questions, please contact support.


How to use Options.txt


First you have to create a text file called "Options.txt" in the same folder where Live's Preferences.cfg file is located. Here's where to find the preferences folder (Live 6.0.9 or higher):


Windows XP
\Documents and Settings\[username]\Application Data\Ableton\Live x.x.x\Preferences\
Windows Vista, Windows 7, Windows 8
\Users\[username]\AppData\Roaming\Ableton\Live x.x.x\Preferences\
Mac OS X
/Users/[username]/Library/Preferences/Ableton/Live x.x.x/
Please refer to the following knowledge base article for further information on how to access hidden folders on Windows and Mac OSX.


You can activate several options in the Options.txt file but use only one per line. The syntax is very simple; each option begins with a hyphen and some options also require an additional underscore (see examples below). If you use the wrong syntax you will get an error message when opening Live. You need to restart Live to activate the option. The file format of the Options.txt must be plain text.


"ReWireMasterOff"


Disables Live's ReWire master functionality.


Example: -ReWireMasterOff


"ReWireChannels"


Number of ReWire channels appearing in the ReWire master when Live runs as ReWire slave.


Default = 16
Min = 0
Max = 64


You must choose an even number of channels.


Example: -ReWireChannels=32


"PluginAutoPopulateThreshold"


Auto-populates the plugin parameter list if a plugin has a number of less or equal than X parameters


Default = 32
Min = 1
Max = 128


"-1" will always populate the list with max. 128 parameters, regardless how many parameters the plugin has (= similar to Live 7 behaviour)


Supported since: 8.1.1


Example: -_PluginAutoPopulateThreshold=32


"ThinningAggressiveness"


When recording automation data, Live smoothes the recorded data with a pre-defined value. The default value is = 0.45


Higher value = more breakpoints eliminated.


Example: -ThinningAggressiveness=0.1


"DontCombineAPCs"


Deactivate the APC combination mode: don't align and sync the session rings of multiple APCs so they can be moved independently.


Supported since: 8.1.4


Example: -DontCombineAPCs


"AbsoluteMouseMode"


Alternative mouse mode without resetting mouse position - useful for tablets, touch screens, remote desk etc.


Supported since: 8.1


Example: -AbsoluteMouseMode


Note: When enabling Absolute Mouse Mode, some actions that require relative mouse movements may experience problems, eg, fine-tuning a parameter while holding CTRL / CMD. Therefore we would recommend to only enable this option if other steps do not resolve this problem.


"EnsureKeyMessagesForPlugins"


This option solves a problem with certain VST Plugins (such as NI's Reaktor), where keyboard data does not reach the plugin.


Supported since: 7.0.2b3


Note: PC ONLY!


Example: -_EnsureKeyMessagesForPlugins

*sources

https://www.ableton.com/en/articles/optionstxt-file-live/
http://www.ItchyTastyRecords.com

Audiophile Grade Cables. USB, XLR, 1/4, spdif, ADAT, Madi, and all other Music Production Cables.

Here is a list of the Top Audiophile Grade Cables. From USB, XLR, 1/4, spdif, ADAT, Madi, and all other Music Production Cables in between.  These cables are far higher quality than even Mogami and Neutrik cables and connectors. 

Wireworld Cables

Schiit Audio
Audioquest
Cardas Audio
Acoustic Revive
Kimber Audio 
Nordost Audio
Transparent Audio
Kimber Audio
Locus Audio
Oyaide Audio
Furutech
Pangea Audio
Siltech Cables 
Space Shuttle Hi-Tech 
StraightWire
Synergistic Research
WyWires
Ziotek
You can purchase any of these fine cabeles @ http://www.audioadvisor.com  


* Sources 

http://goo.gl/Hfcvrx

http://www.thewelltemperedcomputer.com/HW/USB_cable.htm


http://www.ItchyTastyRecords.com

UCSC ON HEARING AND PERCEPTION 

The operation of the ear has two facets: the behavior of the mechanical apparatus and the neurological processing of the information acquired. The mechanics of hearing are straightforward and well understood, but the action of the brain in interpreting sounds is still a matter of dispute among researchers.



Fig. 1 Parts of the ear

1. Auditory canal 6. Round window
2. Ear drum  7. Oval window
3. Hammer  8. Semicircular canals
4. Anvil  9. Cochlea
5. Stirrup  10. Eustachian tube

THE EAR MECHANISM

The ear contains three sections, the outer, middle, and inner ears. The outer ear consists of the lobe and ear canal, structures which serve to protect the more delicate parts inside.
The outer boundry of the middle ear is the eardrum, a thin membrane which vibrates in sympathy with any entering sound. The motion of the eardrum is transferred across the middle ear via three small bones named the hammer, anvil, and stirrup. These bones are supported by muscles which normally allow free motion but can tighten up and inhibit the bones' action when the sound gets too loud. The leverages of these bones are such that rather small motions of the ear drum are very efficiently transmitted.
The boundry of the inner ear is the oval window, another thin membrane which is almost totally covered by the end of the stirrup. The inner ear is not a chamber like the middle ear, but consists of several tubes which wind in various ways within the skull. Most of these tubes, the ones called the semicircular canals, are part of our orientation apparatus. (They contain fine particles of dust-the location of the dust tells us which way is up.) The tube involved in the hearing process is wound tightly like a snail shell and is called the cochlea.


Fig 2. Schematic of the ear
This is a diagram of the ear with the cochlea unwound. The cochlea is filled with fluid and is divided in two the long way by the basilar membrane. The basilar membrane is supported by the sides of the cochlea but is not tightly stretched. Sound introduced into the cochlea via the oval window flexes the basilar membrane and sets up traveling waves along its length. The taper of the membrane is such that these traveling waves are not of even amplitude the entire distance, but grow in amplitude to a certain point and then quickly fade out. The point of maximum amplitude depends on the frequency of the sound wave.
The basilar membrane is covered with tiny hairs, and each hair follicle is connected to a bundle of nerves. Motion of the basilar membrane bends the hairs which in turn excite the associated nerve fibers. These fibers carry the sound information to the brain. This information has two components. First, even though a single nerve cell cannot react fast enough to follow audio frequencies, enough cells are involved that the aggregate of all the firing patterns is a fair replica of the waveform. Second, and probably most importantly, the location of the hair cells associated with the firing nerves is highly correlated with the frequency of the sound. A complex sound will produce a series of active loci along the basilar membrane that accurately matches the spectral plot of the sound.
The amplitude of a sound determines how many nerves associated with the appropriate location fire, and to a slight extent the rate of firing. The main effect is that a loud sound excites nerves along a fairly wide region of the basilar membrane, whereas a soft one excites only a few nerves at each locus.

PERCEPTION

The mechanical process described so far is only the beginning of our perception of sounds. The mechanisms of sound interpretation are poorly understood, in fact is not yet clear whether all people interpret sounds in the same way. Until recently, there has been no way to trace the wiring of the brain, no way to apply simple stimuli and see which parts of the nervous system respond, at least not in any detail. The only research method available was to have people listen to sounds and describe what they heard. The variability of listening skills and the imprecision of the language combined to make psycho-acoustics a rather frustrating field of study. Some of the newest research tools show promise of improving the situation, so research that is happening now will likely clear up several of the mysteries.
The current best guess as to the neural operation of hearing goes like this:
We have seen that sound of a particular waveform and frequency sets up a characteristic pattern of active locations on the basilar membranes. (We might assume that the brain deals with these patterns in the same way it deals with visual patterns on the retina.) If a pattern is repeated enough we learn to recognize that pattern as belonging to a certain sound, much as we learn a particular visual pattern belongs to a certain face. (This learning is accomplished most easily during the early years of life.) The absolute position of the pattern is not very important, it is the pattern itself that is learned. We do possess an ability to interpret the location of the pattern to some degree, but that ability is quite variable from one person to the next. (It is not clear whether that ability is innate or learned.) What use the brain makes of the fact that the aggregate firing of the nerves more or less approximates the waveform of the sound is not known. The processing of impulse sounds (which do not last long enough to set up basilar patterns) is also not well explored.

INTERPRETATION OF SOUNDS

Most studies in psycho-acoustics deal with the sensitivity and accuracy of hearing. This data was intended for use in medicine and telecommunications, so it reflects the abilities of the average untrained listener. It seems to be traditional to weed out musicians from such studies, so the capabilities of trained ears are not documented. I suspect such capabilities are much better than that suggested by the classic studies.

LOUDNESS

The ear can respond to a remarkable range of sound amplitude. (Amplitude corresponds to the quality known as loudness.) The ratio between the threshold of pain and the threshold of sensation is on the order of 130 dB, or ten trillion to one. The judgment of relative sounds is more or less logarithmic, such that a tenfold increase in sound power is described as "twice as loud". The just noticeable difference in loudness varies from 3 dB at the threshold of hearing to an impressive 0.5 dB for loud sounds.

Fig. 3 Perceived loudness of sounds
The sensation of loudness is affected by the frequency of the sound. A series of tests using sine waves produces the curves shown. At the low end of the frequency range of hearing, the ear becomes less sensitive to soft sounds, although the pain threshold as well as judgments of relatively loud sounds are not affected much. Sounds of intermediate softness show some but not all of the sensitivity loss indicated for the threshold of hearing. At high frequencies the change in the sensitivity is more abrupt, with sensation ceasing entirely around 20 khz. The threshold of pain increases in the top octave also.
The ability to make loudness judgments is compromised for sounds of less than 200ms duration. Below that limit, the loudness is affected by the length of the sound; shorter is softer. Durations longer than 200ms do not affect loudness judgment, beyond the fact that we tend to stop paying attention to long unchanging tones.

MASKING

The threshold of hearing for a particular tone can be raised by the presence of another noise or another tone. White noise reduces the loudness of all tones, regardless of absolute level. If the bandwidth of the masking noise is reduced, the effect of masking loud tones is reduced, but the threshold of hearing for those tones remains high. If the masking sound is narrow band noise or a tone, masking depends on the frequency relationship of the masked and masking tones. At low loudness levels, a band of noise will mask tones of higher frequency than the noise more than those of lower frequency. At high levels, a band of noise will also mask tones of lower frequency than itself.

PITCH

People's ability to judge pitch is quite variable. (Pitch is the quality of sound associated with frequency.) Most subjects studied could match pitches very well, usually getting the frequencies of two sine waves within 3%. (Musicians can match frequencies to 1%, or should be able to.) Better results are obtained if the stimuli are similar complex tones, which makes sense since there are more active points along the basilar membrane to give clues. Dissimilar complex tones are apparently fairly difficult to match for pitch (judging from experience with ear training students; I haven't seen any studies on the matter to compare them with sine tone results).
Judgment of relative pitch intervals is extremely variable. The notion of the two to one frequency ratio for the octave is probably learned, although it is easily learned given access to a musical instrument. An untrained subject, asked to set the frequency of a tone to twice that of a reference, is quite likely to set them a twelfth or two octaves apart or find some arbitrary and inconsistent ratio. The tendency to land on "proper" intervals increases if complex tones are used instead of sine tones. Trained musicians often produce octaves slightly wider than two to one, although the practical aspects of their instrument strongly influence their sense of interval. (As a bassoonist who has played the same instrument for twenty years, I have a very strong tendency to place G below middle C a bit high.)
Identification of intervals is even more variable, even among musicians. It does appear to be trainable, suggesting it is a learned ability. Identification of exact pitches is so rare that it has not been properly studied, but there is some anecdotal evidence (such as its relatively more common occurrence among people blind from birth) suggesting it is somehow learned also.
The amplitude of sound does not have a strong effect on the perception of pitch. Such effects seem to hold only for sine tones. At low loudness levels pitch recognition of pure tones becomes difficult, and at high levels increasing loudness seems to shift low and middle register pitches down and high register pitches up.
The assignment of the quality of possessing pitch in the first place depends on the duration and spectral content of the sound. If a sound is shorter than 200ms or so, pitch assignment becomes difficult with decreasing length until a sound of 50ms or less can only be described as a pop. Sounds with waveforms fitting the harmonic pattern are clearly heard as pitched, even if the frequencies are offset by some additive factor. As the spectral plot deviates from the harmonic model, the sense of pitch is reduced, although even noise retains some sense of being high or low.

TIMBRE

Recognition of sounds that are similar in aspects other than pitch and loudness is not well studied, but it is an ability that everyone seems to share. We do know that timbre identification depends strongly on two things, waveform of the steady part of the tone, and the way the spectrum changes with time, particularly at the onset or attack. This ability is probably built on pattern matching, a process that is well documented with vision. Once we have learned to identify a particular timbre, recognition is possible even if the pitch is changed or if parts of the spectrum are filtered out. (We are good enough at this that we can tell the pitch of low sounds when played through a sound system that does not reproduce the fundamentals.)

LOCALIZATION

We are also able to perceive the direction of a sound source with some accuracy. Left and right location is determined by perception of the difference of arrival time or difference in phase of sounds at each ear. If there are more than two arrivals, as in a reverberant environment, we choose the direction of the first sound to arrive, even if later ones are louder. Localization is most accurate with high frequency sounds with sharp attacks.
Height information is provided by the shape of our ears. If a sound of fairly high frequency arrives from the front, a small amount of energy is reflected from the back edge of the ear lobe. This reflection is out of phase for one specific frequency, so a notch is produced in the spectrum. The elongated shape of the lobe causes the notch frequency to vary with the vertical angle of incidence, and we can interpret that effect as height. Height detection is not good for sounds originating to the side or back, or lacking high frequency content.


*Sources*
http://artsites.ucsc.edu/EMS/Music/tech_background/TE-03/teces_03.html
http://www.ItchyTastyRecords.com