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The main user interface is organized into five sections: scenes, sources, audio mixer, transitions, and controls. Scenes are groups of sources like live and recorded video, text and audio. The mixer panel lets the user mute the audio, and adjust the volume through virtual faders, and apply effects by pressing the cogwheel next to the mute ...
The simplest way to change the duration or pitch of an audio recording is to change the playback speed. For a digital audio recording, this can be accomplished through sample rate conversion. When using this method, the frequencies in the recording are always scaled at the same ratio as the speed, transposing its perceived pitch up or down in ...
ASIO 2.3 introduced monitoring for dropouts in the audio stream. [4] ASIO bypasses the normal audio path from a user application through layers of intermediary operating system software so that an application connects directly to the sound card hardware. Each layer that is bypassed means a reduction in latency (the delay between an application ...
Line level is the specified strength of an audio signal used to transmit analog sound between audio components such as CD and DVD players, television sets, audio amplifiers, and mixing consoles. Generally, line level signals sit in the middle of the hierarchy of signal levels in audio engineering.
A stereo microphone is often used for broadcast applications or field recording where it would be impractical to configure two separate condenser microphones in a classic X-Y configuration (see microphone practice) for stereophonic recording. Some such microphones have an adjustable angle of coverage between the two channels.
Real-Time Messaging Protocol (RTMP) is a communication protocol for streaming audio, video, and data over the Internet. Originally developed as a proprietary protocol by Macromedia for streaming between Flash Player and the Flash Communication Server, Adobe (which acquired Macromedia) has released an incomplete version of the specification of ...
In audio compression, audio samples are processed using an audio codec. In a lossless codec audio samples are processed without discarding information by packing repetitive or redundant samples into a more efficiently stored form. A lossless decoder then reproduces the original PCM with no change in quality.
Core Audio is a low-level API for dealing with sound in Apple's macOS and iOS operating systems.It includes an implementation of the cross-platform OpenAL. [1]Apple's Core Audio documentation states that "in creating this new architecture on Mac OS X, Apple's objective in the audio space has been twofold.