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  • 5.1 Surround
    5.1 Surround
    This describes a particular layout of loudspeaker systems with the following channels:

    Channel 1: FRONT LEFT
    Channel 2: FRONT RIGHT
    Channel 3: SURROUND LEFT
    Channel 4: SURROUND RIGHT
    Channel 5: CENTER
    (on-screen dialog)

    (Those are the first 5 channels in "5.1.")

    Channel 6: LFE or SUBWOOFER

    (That's the ".1" in 5.1)


    A "5.1 amplifier" has the appropriate channels of amplification and other controls for driving a 5.1 surround speaker setup.
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  • A2DP Advanced Audio Distribution Profile

    Advanced Audio Distribution Profile (A2DP)

    This profile defines how high quality audio (stereo or mono) can be streamed from one device to another over a Bluetooth connection.[1] For example, music can be streamed from a mobile phone, to a wireless headset, hearing aid & cochlear implant streamer, or car audio or from a laptop/desktop to a wireless headset.
     
    A2DP was initially used in conjunction with an intermediate Bluetooth transceiver that connects to a standard audio output jack, encodes the incoming audio to a Bluetooth-friendly format, and sends the signal wirelessly to Bluetooth headphones that decode and play the audio. Bluetooth headphones, especially the more advanced models, often come with a microphone and support for the Headset (HSP), Hands-Free (HFP) and Audio/Video Remote Control (AVRCP) profiles.
     
    A2DP is designed to transfer a uni-directional 2-channel stereo audio stream, like music from an MP3 player, to a headset or car radio.[2] This profile relies on AVDTP and GAVDP. It includes mandatory support for the low-complexity SBC codec (not to be confused with Bluetooth's voice-signal codecs such as CVSDM), and supports optionally: MPEG-1 , MPEG-2, MPEG-4, AAC, and ATRAC, and is extensible to support manufacturer-defined codecs, such as apt-X. Some Bluetooth stacks enforce the SCMS-T digital rights management (DRM) scheme. In these cases, it is impossible to connect certain A2DP headphones for high quality audio.
  • Analog Audio

    Analog Audio

    It's easiest to describe what Analog Audio is by example.

    1. A saxophonist plays a note in a smoky basement jazz club. The vibrating air coming from the horn moves the air in the smoky room, and your eardrums vibrate back and forth along with the vibration of the air molecules. We experience these vibrations as "sound."
    2. A microphone and an analog tape recorder are set up in the room. The saxophone vibrates the air around it, setting up a series of pressure changes that radiate through the air in the room. When these pressure changes reach the microphone's diaphragm, it shakes back and forth with the vibrations, much like the tympanic membranes in our ears. The microphone "hears" these vibrations and converts them into electrical voltages that are an "analogy" of the air pressure changes that made the original sounds. The tape recorder's record head then stores these electrical voltages ("analog audio signal") on magnetic tape as magnetic fluctuations.
    3. After the set is over, we take the tape recorder home and hook it up to our stereo system. Now we can play the recording back. We play the tape, the magnetic fluctuations on the analog tape are converted to electrical voltage changes (analog audio signal) by the tape playback head and the resulting voltages are sent to our stereo amplifier. The amplifier changes those fluctuating voltages into larger current fluctuations which move our stereo speakers back and forth, far and fast enough to create disturbances in the air of our listening room. If the new air fluctuations coming from our stereo are almost exactly the same as the original vibrations caused by the saxophone playing in the jazz club then we have true High Fidelity analog audio
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  • ASIO

    ASIO


    Brought to you by Steinberg , ASIO (Audio Stream Input/Output) is a driver layer designed for fast, low-latency multichannel audio I/O. It's way superior to anything Windows will give you (WDM, DirectX, etc.), but it's meant for pro audio use. No TB cards support it, nor will they.

    Wikipedia definition:
    http://en.wikipedia.org/wiki/Audio_Stream_Input/Output

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  • Bit Depth and Sampling Rate - Resolution

    Bit Depth and Sampling Rate - Resolution

    RESOLUTION FOR PULSE CODE MODULATION (PCM) DIGITAL RECORDING:

    Bit depth describes how many 0's and 1's can be stored in each snapshot (sample) taken to digitally record each instant of the incoming audio signal. The more bits you use in each sample, the more information each sample can hold, and the higher resolution the digital recording will be in the end. A Compact Audio disc ("CD") stores audio data in 16 bit samples. The bit depth is then described as "16 bits." The latest Hi-Definition PCM digital audio recorders can save audio data at up to 32 bits per sample.

    Sampling Rate describes how many times per second a sample is taken of the incoming audio to be digitally recorded. (Each sample can be any number of bits.) The more samples per second (the "higher the sampling rate") the higher the resolution of the final digital recording. [Note: kHz = thousands of cycles per second]

    The Resolution of the digital recording is the combination of the bit depth and sampling rate. CD plays back its 16-bit digital audio samples at 44,100 times per second. The resolution of CD audio is then described as"16-bit/44.1kHz". That's 16-bit samples taken 44,100 times per second.

    CD QUALITY = 16 bit/44.1kHz
    HIGH-RES DVD AUDIO = 24 bit/96kHz
    HIGH-DEF AUDIO = 32 bit/192kHz



    DIRECT STREAM DIGITAL (DSD)
    Direct Streaming Digital (DSD) is a form of digital recording that, unlike PCM, takes only a one bit sample. Each "0" sample means the voltage to be described goes down, while each "1" sample means the voltage to be described goes up. The sampling rate is super-fast: 2.8224 MHz (millions of samples per second). The stream is continuous, just like analog audio. That's reputed to be the advantage of DSD over PCM, as PCM by definition chops up the audio stream into individual samples. The resolution of DSD is approximately equivalent to that of Hi-Def audio, 32-bit/192kHz.

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  • Blu-ray

    Blu-ray Disc
    (also known as BD or Blu-ray) is an optical disc storage medium designed to supersede the standard DVD format. Its main uses are for storing high-definition video, PlayStation 3 video games, and other data, with up to 25 GB per single layered, and 50 GB per dual layered disc. Although these numbers represent the standard storage for Blu-Ray drives, the specification is open-ended, with the upper theoretical storage limit left unclear. 200 GB discs are available, and 100 GB discs are readable without extra equipment or modified firmware. The disc has the same physical dimensions as standard DVDs and CDs.


    Blu-ray Disc
    Blu-ray Disc



    NOTES:

    • Blu-ray format was brought to you by Sony.
    • PS3 supports Blu-ray discs. PS3 games come on Blu-ray discs.
    • Movies on Blu-ray discs often include the latest HiDef audio formats such as dts MA (MasterAudio). Blu-ray movies can also include a Dolby TrueHD soundtrack, or they can also include a Dolby Digital 5.1 or standard dts soundtrack (lower quality, more widely compatible).
    • Microsoft Xbox 360 does not support Blu-ray discs at all (booooo).

    Our X41 and DSS do not decode dts of any kind, or Dolby TrueHD. If you want to play a Blu-ray disc's surround soundtrack in our surround products, choose the Dolby Digital 5.1 or EX track and connect Tslink cable from PS3 or Blu-ray player's optical Digital Out to optical Digital In on our product.
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  • DC Offset

    DC Offset


    The Wikipedia definition:http://en.wikipedia.org/wiki/DC_offset

    The definition as it pertains to sound cards:

    When a digital audio recording is made, the incoming sound signal is converted from varying voltages (analog audio) to 0's and 1's (binary code or digital audio).

    It is assumed that the "ground" or "earth" of the incoming audio is at 0 volts AC. That way, when a "0" state is recorded in the digital audio, the system knows it's supposed to be at zero volts (no signal at all).

    If for some reason there's a problem with the input amplifier in the digital recorder, and that problem raises the ground voltage to a slight positive voltage (say +0.5 V) or a slight negative voltage (let's say -0.5V), then the zero state will actually be at a voltage other than zero. This is not a good thing.

    The problem is that when you try to edit the resulting audio recording, a part of the file will be joined to another. When this happens, the no-signal part of the file will not be at zero volts, but at some other voltage. Audio editing programs will always try to join together pieces of files at the zero crossings of the audio waveforms, in order to eliminate the pop or click noises that would occur if the two audio sections were mismatched. The DC offset will actually force a mismatch, however, and editing a file with a DC offset will create a click or pop noise wherever sections of the audio data have been joined together. This is not good, as audio programs are always starting and stopping, or changing the audio data on the fly. You don't want clicks and pops in everything you play, or every time you pause or start your audio recordings.

    How can you tell if there is a DC offset in a signal? In your favorite audio editing software:
    1. Record a few seconds of absolute silence to a digital audio file (e.g. a WAV file).
    2. Open the resulting recording and look at its waveform display.
    3. There should be a "zero line" in the middle of the display. The center of the waveform should be on that "zero line."

    If you notice that the waveform's center is at a level either above or below the "zero line," then there is a DC Offset in the recorded audio.
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  • Dolby Digital (5.1, 7.1)

    Dolby Digital 5.1 and 7.1 (incl. Dolby Digital EX)


    TheWikipedia definition for Dolby Digitalprobably tells you everything you'll ever need to know.

    Don't forget to visit the Dolby Labs website , especially their area for Professional information .
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  • DTS (Digital Theater Systems)

    DTS (Digital Theater Systems)


    One visit to the DTS websitewill probably answer just about any questions you might have. Suffice to say that DTS is very similar to Dolby Digital, except a bit better overall.

    DVD movie discs usually have a Dolby Digital soundtrack, but some use DTS encoding.

    Blu-Ray movie discs usually have a DTS Master or similar soundtrack, but may also include a Dolby TrueHD layer.
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  • DVD Video

    DVD Video


    Basically, this is a 4.7 GB sized version of the Compact Disc. A DVD is large enough to hold the video data for a feature-length movie plus its surround soundtrack (when compressed and encoded into Dolby Digital or DTS audio).

    For our purposes, a DVD movie disc will have the standard definition video data of a movie plus its movie soundtrack in Dolby Didgital or DTS format.
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  • Echo

    Echo

    In audio signal processing and acoustics, an echo (plural echoes) is a reflection of sound, arriving at the listener some time after the direct sound. Typical examples are the echo produced by the bottom of a well, by a building, or by the walls of an enclosed room. A true echo is a single reflection of the sound source. The time delay is the extra distance divided by the speed of sound.

    Acoustic phenomenon (Advanced)
    :
    If so many reflections arrive at a listener that they are unable to distinguish between them, the proper term is reverberation. An echo can be explained as a wave that has been reflected by a discontinuity in the propagation medium, and returns with sufficient magnitude and delay to be perceived. Echoes are reflected off walls or hard surfaces like mountains and privacy fences.
     
    When dealing with audible frequencies, the human ear cannot distinguish an echo from the original sound if the delay is less than 1/10 of a second. Thus, since the velocity of sound is approximately 343 m/s at a normal room temperature of about 20°C, the reflecting object must be more than 17.15 m from the sound source at this temperature for an echo to be heard by a person at the source.
     
    Sound travels approximately 343 metres/s (1100 ft/s). If a sound produces an echo in 2 seconds, the object producing the echo would be half that distance away (the sound takes half the time to get to the object and half the time to return). The distance for an object with a 2-second echo return would be 1 sec X 343 metres/s or 343 metres (1100 ft). In most situations with human hearing, echoes are about one-half second or about half this distance, since sounds grow fainter with distance. In nature, canyon walls or rock cliffs facing water are the most common natural settings for hearing echoes.The strength of an echo is frequently measured in dB sound pressure level SPL relative to the directly transmitted wave. Echoes may be desirable (as in sonar) or undesirable (as in telephone systems).
  • Encode - Decode - CODEC

    Encode - Decode - CODEC


    A "codec" is an encoding/decoding scheme, system or "algorithm." (EnCOde-DECode). Think of it like a Morse code transmission:

    1. You start with a sentence you've written in English.
    2. You then ENCODE the letters of the words into dots and dashes.
    3. You transmit the resulting Morse code (sequence of dots and dashes).
    4. On the receiving end, the operator will DECODE the Morse code back into English and be able to read the message.

    Mission accomplished. You could think of Morse Code as the "codec" you used to perform the above. So it is with digital audio codecs like MP3, FLAC, dts and Dolby Digital.

    Now let's take a DVD movie as our example.

    1. You start with a movie soundtrack made of the six separate audio channels that make a 5.1 playback scheme.
    2. Using Dolby Digital 5.1, we will ENCODE the six channels onto the two channels of a S/PDIF data stream. The encoded digital audio is included on a DVD movie disc. [Note: S/PDIF can only carry two channels of audio data, so the six 5.1 surround channels have to be encoded to fit onto two channels to be sent over S/PDIF.]
    3. Using TOSlink optical S/PDIF, the DVD player sends the Dolby Digital-encoded audio stream to your home theater receiver's optical S/PDIF input, and on to the receiver's Dolby Digital decoder.
    4. The home theater receiver's Dolby Digital decoder will DECODE the incoming data stream, separating out the six original channels. Then the receiver's digital-to-analog converters will send the six separate analog audio channels to six separate amplifiers and finally to six separate speakers.
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  • Feedback

    Feedback


    Generally speaking, when the term feedback is used in audio, it's usually meant to mean positive feedback.

    The most familiar example of positive feedback is when:
    1. You have a microphone (mic) connected to the input of an audio amplifier feeding a  loudspeaker (speakers).
    2. As long as the mic is located far enough away from the speaker(s), when you speak into the mic your voice will be amplified nice and clearly through the amp and speaker.
    3. If, however, you put the microphone right up against the speaker, the background noise of the room as picked up by the microphone (and the self-noise of the mic and amp) will play through the speaker, which will then be picked up by the microphone, and the whole thing will be amplified (made louder) by the amplifier, which will then play louder through the speaker, which will again be picked up by the microphone, and again amplified and played back into the microphone, and 'round and 'round until this vicious cycle of highly amplified self-noise creates a continuous howl and/or screech playing through the speaker. This is described as feedback.

    The output from the system is fed back into the input of the system, which increases the level of the signal coming in the input, which increases the output and again increases the level at the input, until the system goes into self-oscillation (makes its own signal instead of only responding to an outside signal applied to its input). Feeding a portion of the output back into the input increases the sensitivity of the system.

    Negative feedback is when the output is fed back into the input in such a way that it reduces the sensitivity of the system. This is a useful concept in the design of audio amplification and other electronic circuits, but is not something you'll encounter much in the world of everyday physical objects.
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  • Frequency Response

    Frequency response

    Frequency response is the quantitative measure of the output spectrum " often the amplitude response " of a system or device in response to a stimulus; in electrical terms this stimulus would be an input signal. In the audible range it is usually referred to in connection with electronic amplifiers, microphones and loudspeakers. Radio spectrum frequency response can refer to measurements of coaxial cable, twisted-pair cable, video switching equipment, wireless communications devices, and antenna systems. Infrasonic frequency response measurements include earthquakes and electroencephalography (brain waves).

    Frequency response requirements differ depending on the application. In high fidelity audio, an amplifier requires a frequency response of at least 20-20,000 Hz, with a tolerance as tight as ±0.1 dB in the mid-range frequencies around 1000 Hz, however, in telephony, a frequency response of 400-4,000 Hz, with a tolerance of ±1 dB is sufficient for intelligibility of speech.

    Frequency response curves are often used to indicate the accuracy of electronic components or systems. When a system or component reproduces all desired input signals with no emphasis or attenuation of a particular frequency band, the system or component is said to be "flat", or to have a flat frequency response curve.

  • Ground (electrical)

    Ground (electrical)


    The Wikipedia definition:
    http://en.wikipedia.org/wiki/Ground_(electricity)

    "In electrical engineering, ground or earth may be the reference point in an electrical circuit from which other voltages are measured, or a common return path for electric current, or a direct physical connection to the Earth."

    Usually ground is set at zero (0) volts, and is connected (at least indirectly) to Earth, e.g. by electrical wiring that eventually connects to a conductive metal post stuck into the ground. The Earth is at 0 volts, so connecting something to the electrical ground ("grounding" it, or making a "grounded" connection) will set that connection to 0 volts as well.

    A good way to visualize how an electrical ground works is to think about a lightning rod on the roof of a house. A metal rod sticks up higher than anything else on the roof, so that lightning will be most likely to hit that (and not something inside the house). A stout, conductive wire runs from the lightning rod, down the side of the house and deep into the ground. If a bolt of lightning hits the lightning rod, the current from the lightning will flow straight through the rod and wire, down into the ground -- the lightning's electricity is "grounded" (shunted to ground) so that it doesn't pass through the house (or its occupants).

    In an electronic circuit, there will be many connections that should go to 0 volts, which all parts of the electrical system will have in common. This is the common ground (may be abbreviated as COM), which you may see labeled as such on the speaker connectors of older stereo amplifiers and loudspeakers. The circuits' individual grounding points will all be connected to this common ground, which will then be connected to the reference ground (earth). In this way, all individual circuits within a device can be assured of proper grounding.
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  • Headsets and Headphones -- What's the Difference?

    Headsets and Headphones -- What's the Difference?

    Headphones (or a headphone set) is a device with speaker enclosures making two earcups, one for each ear. Stereo sound can be played through the headphones, with the left channel program playing into the left ear, and the right channel program playing into the right ear.

    A Headset is a pair of headphones with a microphone "boom" added. Out in the workaday world, you'll see headsets used by helicopter and airplane pilots such as: Aviator headset (microphone attached from left earcup)


    We make headsets for PC and console gaming.

    Turtle Beach X11 Headset (mic attached from left earcup)
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  • Hertz, Hz (cycles per second)

    Hertz, Hz (cycles per second)


    First, the Wikipedia definition:
    http://en.wikipedia.org/wiki/Hertz

    "The hertz (symbol: Hz) is the SI (International System of Units) unit of frequency defined as the number of cycles per second of a periodic phenomenon. One of its most common uses is the description of sinusoidal waves, particularly those used in radio and audio applications."

    A sound wave is a vibration which occurs at a certain frequency. The lower the frequency, the lower the pitch of the sound. The higher the frequency the higher the pitch.

    The number of "vibrations per second" is expressed as "cycles per second" (cps) or Hertz (Hz).

    Thousands of cycles per second is abbreviated to kHz or "kilohertz" (k = a thousand, Hz = cycles per second).

    Millions of cycles per second is abbreviated to MHz or "megahertz" (M = one million, Hz = cycles per second).

    Billions of cycles per second is abbreviated to GHz or "gigahertz" (G = one billion, Hz = cycles per second).

    Hz - (Hertz x 1)
    kHz - (Hz x 1000, or kilohertz)
    MHz - (Hz x 1,000,000 or megahertz)
    GHz - (Hz x 1,000,000,000 or gigahertz)
    THz - (Hz x 1,000,000,000,000 or terahertz)

    Audible sound is defined as sound vibrations occurring at frequencies of 20 Hz up to 20 kHz.



    Radio frequencies vibrate at a faster rate than do sound waves in the audible band, so radio frequencies can be defined as occurring from about several hundred kHz up into the GHz range.

    Some real world examples:
    • 40 Hz is about where the lowest note is on a bass guitar or bass viol (low "E"). It's also about the lowest tone in a bass drum.
    • 80 Hz is about where the lowest note is on a guitar (low "E") .
    • 2122 Hz (2.122 kHz) is the highest note on a piano.
    • 16 kHz is about the highest tone a typical 40 year old human male can hear.
    • 530 kHz is about the lowest frequency on the AM radio band. 1600 kHz (1.6 MHz) is about the highest AM radio frequency.
    • 88 MHz is about the lowest frequency on the FM radio band. 107 MHz is about the highest FM radio frequency,
    • 2.4 GHz is the frequency band used for transmission/reception in most wireless consumer electronics (RF wireless).



    Infrared frequencies are very, very high. Infrared is far higher in frequency than radio frequencies, but not quite as high in frequency as visible light. Somebody on Wikipedia explained it like this:

    "Infrared radiation (IR) is electromagnetic radiation with a wavelength between 0.7 and 300micrometres, which equates to a frequency range between approximately 1 and 430 THz.[1]
    Its wavelength is longer (and the frequency lower) than that of visible light, but the wavelength is shorter (and the frequency higher) than that of terahertz radiation microwaves."


    Check out the Wikipedia Infrared entry for more info:
    http://en.wikipedia.org/wiki/Infrared
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  • HISS (What is?)

    HISS (What is?)

    In audio, "hiss" describes a continuous broadband noise of fairly even amplitude (level). There is often a tiny bit of hiss audible in the output of any amplifier or audio device that incorporates amplifiers (like our amplified gaming headsets).

    Hiss is best described as the sound of a distant waterfall or the sound of a gentle wind that never varies. If you're familiar with old-fashioned analog cassette tapes, imagine the sound you hear when you switch off the Dolby B noise reduction. That's hiss.

    By comparison, "static" is another broadband noise, but of varying amplitude that sounds more the like the sound of frying bacon, complete with intermittent popping and/or crackling noises.

    Customers often describe noises they hear in inconsistent ways, referring to static when they probably mean hiss. It's important to get them to define the noise so that we can understand it, as excessive static or hum in a product will usually indicate a more serious problem than a little bit of hiss in the background.
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  • Hum

    Hum

    "Hum" is an electronic noise (unwanted output from a circuit). It usually sounds like a fairly smooth, continuous low tone at about 120 Hz. It can also occur at an even lower tone, 60 Hz.

    In most power supply designs, the 60 Hz frequency of the 117 VAC mains (the electricity you get from a wall outlet, or the "wall juice.") needs to be filtered out. Power supply "filter" circuits block this 60 Hz AC component (called "ripple") by diverting it to ground ("earth") so that it is not passed on to the DC circuits downstream and mixed into the desired signal(s).

    Hum is usually the result of this AC power supply "ripple" noise leaking into the DC circuits of an audio amplifier. This can be caused by inadequate power supply design or by a broken or defective part in the power supply.

    Hum can also occur if an audio cable has a disconnected ground wire, or if a plug or jack has a damaged ground connection.

    A "ratty buzz" is actually different than a hum, both in how it sounds and in what causes it.

    The buzz noise will be of an indefinite pitch, or contain a number of combined pitches. It will sound more fuzzy and rough in sonic "texture." While a simple hum usually indicates a power supply problem, a buzz might indicate a ground loop, where the signal ground connections of two or more devices have become connected in an undesirable way. A ground loop can be interrupted by unplugging one of the offending devices, or by connecting the various devices in a different way that does not create the loop through which the signal ground is incorrectly connected. (See "Ground Loop")
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  • IEEE 802.11 g/b/n

    IEEE 802.11 g/b/n


    Wiki: http://en.wikipedia.org/wiki/IEEE_802.11

    • 802.11g (Wireless "G") was the most common iteration until about 2008.
    • 802.11n (Wireless "N") uses more bandwidth than 802.11g, so often conflicts with our RF wireless headsets.
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  • Latency

    Latency

    Latency refers to a short period of delay (usually measured in milliseconds) between when an audio signal enters and when it emerges from a system. Potential contributors to latency in an audio system include analog-to-digital conversion, buffering, digital signal processing, transmission time, digital-to-analog conversion and the speed of sound in air.

    Latency in computer audio: (Advanced)
    Latency can be a particular problem in current Microsoft Windows audio platforms, but is much less so in Apple's Mac OS X and most Linux operating systems.[dubious - discuss] Mac OS X uses Apple's built-in CoreAudio architecture, which is prepared to run low latencies (as opposed to Windows' WDM architecture). A popular solution is Steinberg's ASIO, which bypasses these layers and connects audio signals directly to the sound card's hardware. Most professional and semi-professional audio applications utilize the ASIO driver, allowing Windows users to work with audio in real time.
     
    With most Linux operating systems, latency tends to be better than with the MME or DirectX drivers of Microsoft Windows, if the modern ALSA sound-architecture is used.
     
    The RT-kernel (RealTime-kernel) is a modified Linux-kernel, that alters the standard timer frequency the Linux kernel uses and gives all processes or threads the ability to have realtime-priority. (This means, that a time-critical process like an audio-stream can get priority over another, less-critical process like network activity. This is also configurable per user (for example, the processes of user "tux" could have priority over processes of user "nobody" or over the processes of several system daemons). On a standard Linux-system, this is only possible with one process at the same time.

  • Microphone ("Mic")

    Microphone ("Mic")

    A microphone is a transducer that changes sound waves (vibrations in air) into electrical signals that can be amplified and then transmitted or recorded.

    A microphone's diaphragm is something like a tiny loudspeaker cone operated in reverse. Sound waves strike the diaphragm and shake it. The diaphragm is connected to a tiny copper coil which moves over a tiny magnet, which induces an electrical signal that is an electrical copy ("analogy" or ANALOG) of the original sound. This electrical current can then be sent to an amplifier and transmitted or recorded.

    An old-fashioned professional broadcast microphone:

    Old style dynamic microphone
    Old style dynamic microphone
  • Microphone Monitoring

    Microphone Monitoring

    This feature allows you to hear your voice in the headset, so you won't have to shout to hear what you're saying.
  • Noise Floor

    Noise Floor

    According to the Wikipedia, "the noise floor is the measure of the signal created from the sum of all the noise sources and unwanted signals within a measurement system."

    The Wikipedia definition:http://en.wikipedia.org/wiki/Noise_floor

    For our purposes, let's look at a sound card's Line Input used to record from a CD player.
    We connect the CD player's analog Line outputs to the Line In on the sound card.
    1. We know the CD player will output a maximum of 2 VAC signal. We set the sound card's Line In level to allow it not to distort (go "in the red") on a 2 V signal peak.
    2. Now we do a test recording, with nothing playing from the CD player.
    We may see that there is a constant noise level in our test recording, at -75 dB (decibels) below the zero (maximum) level. This will be the combined residual hum, hiss and other noise generated by the system (player plus recorder) before any signal is sent from the player to the recorder. That is our "noise floor." We won't be able to record any sounds that are below about 75 dB lower (quieter) than 0 dB ("full-scale," or maximum level, stated as 0 dBFS). If a sound is louder than -75 dB below 0 dBFS, then we'll be able to record it and hear it when we play back the recording. If the sound is quieter than -75 dB below 0 dBFS, then we won't be able to hear it in the recording, as it will be "masked" by the noise ("buried beneath the noise floor").

    The Signal to Noise Ratio could then be said to be -75 dB ("minus seventy-five decibels").
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  • PCI (Peripheral Component Interconnect)

    PCI (Peripheral Component Interconnect)


    From Wikipedia:
    Our Discontinued Montego DDL and Riviera sound cards use the Conventional PCI interface.
    • Both of these products use 5V slots only (will not work with 3.3V PCIe slots).
    • Both of these products are full-height PCI cards. They will not fit in half-height PCI slots.
    We do not make any PCIe or PCI-X sound cards.
    We do not make any half-height PCI cards.
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  • PCIe, PCI Express, PCI-X (see PCI)
    PCIe, PCI Express, PCI-X see PCI (Peripheral Component Interconnect)
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  • PCM (Pulse Code Modulation) - Digital Audio

    PCM - Digital Audio

    While analog audio uses electrical voltage or current fluctuations to replicate the sound waves "heard" by a microphone, digital audio stores these sound waves as a series of 0's and 1's, or in binary code ("digital").

    From http://home.earthlink.net/~rongonz/home_rec/digi_audio_concepts.html

    1. First the original sound is converted to analog audio voltage fluctuations by the microphone(s).
    2. Instead of using an analog tape deck, we are now going to use a digital recorder. Let's use a DAT recorder as our example. The analog audio voltage fluctuations are fed to a circuit called the Analog-to-Digital Converter that changes the incoming voltages to digital "snapshots", 44,100 times a second. Each "snapshot" consists of 16 zeroes and/or ones. Each combination of zeroes and/or ones represents a different signal voltage. Using sixteen 0's and 1's in each "sample", one of 65,536 different voltage levels can be described by each sample. A DAT or CD uses a "sampling rate' of 44,100 samples per second (44.1kHz). This means that 2,890,137,600 different analog audio voltage levels can be described each second -- and you're right, that's a lot. But some say that capturing audio with 16 bits, 44,100 times a second may not be enough to accurately describe what our ears can hear, so that's why there is now a push on to record everything in 24 bits, 96,000 times a second ("24/96 resolution"). The latest digital recorders (such as the new Pro Tools HD system) capture audio at 32 bit, 192kHz resolution, or even a single bit at a time sampled at over 2.1 GHz a second (Direct Stream Digital or "DSD").
    3. When we want to actually hear the digital audio, the audio data has to go through a Digital-to-Analog Converter, which changes the binary code samples to analog voltage fluctuations that are then sent to a power amp and on to the speakers, which shake the air molecules in our listening room enough for us to hear a reasaonably accurate reproduction of the original sound.

    Pulse Code Modulation (PCM) is what we're describing here. PCM audio uses samples that each have a certain number of bits in them, with those samples recorded or played back at a certain number of times per second.

    Resolution:
    The more bits in each sample, the more precisely the sample describes that instant of the sound wave to be captured. The more samples taken per second, the more precisely the entire sound wave can be described. Generally speaking, the bigger the number of bits in each sample and the more frequently samples are taken each second, the higher the resolution of the resulting digital audio recording. _____________________________________________________________________________________________________
  • Piggyback
    RCA Stereo Dual Piggyback Patch Cable


  • RCA Cables


    RCA Jacks:


    An RCA connector, sometimes called a phono connector or cinch connector, is a type of electrical connector commonly used to carry audio and video signals. The name "RCA" derives from the Radio Corporation of America, which introduced the design by the early 1940s to allow mono phonograph players to be connected to amplifiers.

    They began to replace the older TRS connectors (also called jack plugs) for many other applications in the audio world when component high fidelity systems started becoming popular in the 1950s. However, mini TRS connectors (3.5 mm jacks) and sub-miniature (2.5 mm) jacks have been overtaking RCA connectors in some recent applications such as MP3 players.

    RCA Plugs:

    The connection's plug is called an RCA plug or phono plug, for "phonograph". The name "phono plug" is sometimes confused with a "phone plug" which refers to anything from a TRS connector plug to a British phone plug to an RJ14 registered jack plug.


  • Signal to Noise Ratio (S/N)

    Noise Floor


    According to the Wikipedia, "the noise floor is the measure of the signal created from the sum of all the noise sources and unwanted signals within a measurement system."

    The Wikipedia definition: http://en.wikipedia.org/wiki/Noise_floor

    For our purposes, let's look at a sound card's Line Input used to record from a CD player.

    1. We connect the CD player's analog Line outputs to the Line In on the sound card.
    2. We know the CD player will output a maximum of 2 VAC signal. We set the sound card's Line In level to allow it not to distort (go "in the red") on a 2 V signal peak.
    3. Now we do a test recording, with nothing playing from the CD player.

    We may see that there is a constant noise level in our test recording, at -75 dB (decibels) below the zero (maximum) level. This will be the combined residual hum, hiss and other noise generated by the system (player plus recorder) before any signal is sent from the player to the recorder. That is our "noise floor." We won't be able to record any sounds that are below about 75 dB lower (quieter) than 0 dB ("full-scale," or maximum level, stated as 0 dBFS). If a sound is louder than -75 dB below 0 dBFS, then we'll be able to record it and hear it when we play back the recording. If the sound is quieter than -75 dB below 0 dBFS, then we won't be able to hear it in the recording, as it will be "masked" by the noise ("buried beneath the noise floor").

    The Signal to Noise Ratio could then be said to be -75 dB ("minus seventy-five decibels").
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