Search results
Results from the WOW.Com Content Network
When the sound card uses a custom driver for use with the system supplied port class driver PortCls.sys or implements a mini-driver for use with the streaming class driver, applications can bypass the KMixer completely and use the kernel streaming interfaces instead to directly interact with audio driver and reduce latency. Windows 98 includes ...
Most devices capable of compressing audio dynamics can also be used to reduce the volume of one audio source when another audio source reaches a certain level; this is called side-chaining. [13] In electronic dance music , side-chaining is often used on basslines , controlled by the kick drum or a similar percussive trigger, to prevent the two ...
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.
When playing music remotely, musicians must reduce or eliminate the issue of audio latency in order to play in time together. While standard web conferencing software is designed to facilitate remote audio and video communication, it has too much latency for live musical performance.
The ULD is a lossy audio data compression scheme that only introduces a very small amount of delay into the audio signal compared to commonly known audio coders like MP3 or AAC. This property is especially useful for communication purposes (like voice calls, video conferencing or making music via the internet), for which not only high ...
DirectSound is a deprecated software component of the Microsoft DirectX library for the Windows operating system, superseded by XAudio2.It provides a low-latency interface to sound card drivers written for Windows 95 through Windows XP and can handle the mixing and recording of multiple audio streams.
Opus is a lossy audio coding format developed by the Xiph.Org Foundation and standardized by the Internet Engineering Task Force, designed to efficiently code speech and general audio in a single format, while remaining low-latency enough for real-time interactive communication and low-complexity enough for low-end embedded processors.
The Adaptive Multi-Rate (AMR, AMR-NB or GSM-AMR) audio codec is an audio compression format optimized for speech coding. AMR is a multi-rate narrowband speech codec that encodes narrowband (200–3400 Hz) signals at variable bit rates ranging from 4.75 to 12.2 kbit/s with toll quality [ 3 ] speech starting at 7.4 kbit/s.