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The MediaDevices.getUserMedia() method prompts the user for permission to use one video and/or one audio input device such as a camera or screensharing and/or a microphone. If the user provides permission, then the returned Promise is resolved with the resulting MediaStream object. If the user denies permission, or media is not available, then the promise is rejected with PermissionDeniedError or NotFoundError respectively. Note that it is possible for the returned promise to neither resolve nor reject, as the user is not required to make a choice.
API Audio audio Experimental MediaDevices Method Reference Video video


The MediaKeySystemConfiguration.audioCapabilities read-only property returns an array of supported audio type and capability pairs.
API Audio audio EncryptedMediaExtensions initDataTypes Media MediaKeySystemConfiguration NeedsExample Property Reference

OfflineAudioContext.startRendering (promise)

The promise-based startRendering() method of the OfflineAudioContext Interface starts rendering the audio graph, taking into account the current connections and the current scheduled changes.
API Audio Method OfflineAudioContext Promise Reference startRendering Web Audio API


The pan property takes a unitless value between -1 (full left pan) and 1 (full right pan). This interface was introduced as a much simpler way to apply a simple panning effect than having to use a full PannerNode.
API Audio Interface Reference StereoPannerNode Web Audio API

Improving compatibility using WebRTC adapter.js

While the WebRTC specification is relatively stable, not all browsers have fully implemented all of its features. In addition, some browsers still have prefixes on some or all WebRTC APIs. While you can manually code around these issues, there is an easier way. The WebRTC organization provides on GitHub the WebRTC adapter to work around compatibility issues in different browsers' WebRTC implementations. The adapter is a JavaScript shim which lets your code to be written to the specification so that it will "just work" in all browsers with WebRTC support. There's no need to conditionally use prefixed APIs or implement other workarounds.
API Audio Guide Video WebRTC


The createIIRFilter() method of the AudioContext interface creates an IIRFilterNode, which represents a general infinite impulse response (IIR) filter which can be configured to serve as various types of filter.
API audio Audio AudioContext CreateIIRFilter filter Method Reference Web Audio API


The AnalyserNode interface represents a node able to provide real-time frequency and time-domain analysis information. It is an AudioNode that passes the audio stream unchanged from the input to the output, but allows you to take the generated data, process it, and create audio visualizations.
AnalyserNode API Interface Reference Web Audio API


An AudioContext can be a target of events, therefore it implements the EventTarget interface.
API Interface Reference Web Audio API


AudioDestinationNode has no output (as it is the output, no more AudioNode can be linked after it in the audio graph) and one input. The amount of channels in the input must be between 0 and the maxChannelCount value or an exception is raised.
API AudioDestinationNode Interface Reference Web Audio API


There are two kinds of AudioParam, a-rate and k-rate parameters:
API AudioParam Reference Web Audio API


A reference to this AudioParam object. In some browsers older implementations of this interface return void.
API Audio AudioParam Method Reference setValueCurveAtTime Web