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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]
In Windows 95, 98 and Me, the DirectSound mixer component and the sound card drivers were both implemented as a kernel-mode VxD driver (Dsound.vxd), allowing direct access to the primary buffer used by the audio hardware and thus, providing the lowest possible latency between the user-mode API and the underlying hardware, but in some cases ...
A bloated buffer has an effect only when this buffer is actually used. In other words, oversized buffers have a damaging effect only when the link they buffer becomes a bottleneck. The size of the buffer serving a bottleneck can be measured using the ping utility provided by most operating systems. First, the other host should be pinged ...
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 ...
The JACK API is standardized by consensus, and two compatible implementations exist: jack1, which is implemented in plain C and has been in maintenance mode for a while, and jack2 (originally jackdmp), a re-implementation in C++ originally led by Stéphane Letz, which introduced multi-processor scalability and support for operating systems other than Linux.
Latency refers to a short period of delay (usually measured in milliseconds) between when an audio signal enters a system, and when it emerges.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 the transmission medium.
The MPEG-4 Low Delay Audio Coder (a.k.a. AAC Low Delay, or AAC-LD) is audio compression standard designed to combine the advantages of perceptual audio coding with the low delay necessary for two-way communication. It is closely derived from the MPEG-2 Advanced Audio Coding (AAC) standard.
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 ...