Hallo Erik,
Ik krijg de indruk dat je een aantal zaken door elkaar haalt.
Die 192 Khz van de E-mu 1616m heeft niets te maken met de latency, maar alles met de geluidkwaliteit van je track.
Zie onderstaand:
Sampling Rate
When it is necessary to capture audio covering the entire 18-20,000 Hz range of human hearing, such as when recording music or many types of acoustic events, audio waveforms are typically sampled at 44.1 kHz (CD) or 48 kHz (professional audio). The approximately double-rate requirement is a consequence of the Nyquist theorem.
There has been an industry trend towards sampling rates well beyond the basic requirements; 96 kHz and even 192 kHz are available.[1] This is in contrast with laboratory experiments have failed to show that ultrasonic frequencies are audible to human observers, however in some cases ultrasonic sounds do interact with and modulate the audible part of the frequency spectrum (intermodulation distortion). It is noteworthy that intermodulation distortion is not present in the live audio and so it represents an artificial coloration to the live sound.[2]
One advantage of higher sampling rates is that they can relax the low-pass filter design requirements for ADCs and DACs, but with modern oversampling sigma-delta converters this advantage is less important.
(Bron:
http://en.wikipedia.org/wiki/Sampling_(signal_processing))
Latency (audio)
From Wikipedia, the free encyclopedia
(Redirected from Audio latency)
Jump to: navigation, search
Latency refers to a short period of delay (usually measured in milliseconds) required for the conversion between analog and digital representations of the sound data. Devices such as computers can only process digital data. Thus, the analog data it receives on microphone or line-in inputs must be converted to digital data. After processing of data, the processed data must be converted back to an analog signal before it can be output to speakers and played back.
This conversion between analog and digital takes a short amount of time, which is known as latency. Although this process consumes a very small interval, the majority of latency issues arise from the data being handed off by several layers of software, particularly elaborate in current Windows audio platforms and much less so in Apple's OS X and most Linux operating systems. A popular workaround 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, i.e. digital multitrack recording.
One example of latency is a keyboard connected to a computer. When the user hits a key, an audio signal, which is analog, is transferred along the connecting wire in the form of electrical current. The computer would then convert the signal to a digital format and process it according to any settings input by the user. After the processing is complete, the processed digital signal is converted to an analog sound wave (represented by current in the wire), which is then sent to the speaker.
Hoe latency aan te pakken:
http://www.practicalpc.co.uk/computing/sound/latency1.htm
Ik denk dat je, als je een low-latency oplossing wilt, toch het beste voor een FireWire box kunt gaan in combinatie met een FireWire PCMIA kaart. Welke is afhankelijk van je budget en de hoeveelheid i/o's die je nodighebt. Zelf vindt ik dit een mooie deal:
http://www.musik-produktiv.com/shop2/shop04.asp/artnr/100049907/sid/!x03nlnlc/quelle/listen
Krijg je een uitgebreidere versie dan Sonar LE zoals bij de Edirol producten. De box is identiek aan de Edirols.
)
