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Audio Stream Input/Output (ASIO) is a computer audio interface driver protocol for digital audio specified by Steinberg, providing high data throughput, synchronization, and low latency between a software application and a computer's audio interface or sound card. [1]
A popular optimization solution is Steinberg's ASIO, which bypasses the audio platform, and connects audio signals directly to the sound card's hardware. Many professional and semi-professional audio applications utilize the ASIO driver, allowing users to work with audio in real time. [13] Pro Tools HD offers a low latency system similar to ASIO.
While standard web conferencing software is designed to facilitate remote audio and video communication, it has too much latency for live musical performance. [ 1 ] [ 2 ] Connection-oriented Internet protocols subject audio signals to delays and other interference which presents a problem for keeping latency low enough for musicians to play ...
Network latency (link offset) is the time difference between the moment an audio stream enters the source (ingress time), marked by RTP timestamp in the media packet, and the moment it leaves the destination (egress time). Latency depends on packet time, propagation and queuing delays, packet processing overhead, and buffering in the ...
In addition, conferencing software does not normally allow detailed setting of individual audio streams' volume or panning on the user side, both of which are integral features of Jamulus. To reduce latency as much as possible, Jamulus makes use of compressed audio and the UDP protocol to transmit audio data. Total latency is composed of:
Sound Blaster Audigy Player Sound Blaster Audigy 2 ZS Gold. Sound Blaster Audigy is a product line of sound cards from Creative Technology.The flagship model of the Audigy family used the EMU10K2 audio DSP, an improved version of the SB-Live's EMU10K1, while the value/SE editions were built with a less-expensive audio controller.
Windows Vista features a completely re-written audio stack designed to provide low-latency 32-bit floating point audio, higher-quality digital signal processing, bit-for-bit sample level accuracy, up to 144 dB of dynamic range and new audio APIs created by a team including Steve Ball and Larry Osterman.
The scheduling requirements of JACK to achieve sufficiently low latencies were one of the driving forces behind the real-time optimization effort for the Linux kernel 2.6 series, [8] [9] whose initial latency performance had been disappointing compared to the older 2.4 series. [10]