MediaStreamAudioSourceNode - Web APIs 编辑

The MediaStreamAudioSourceNode interface is a type of AudioNode which operates as an audio source whose media is received from a MediaStream obtained using the WebRTC or Media Capture and Streams APIs. This media could be from a microphone (through getUserMedia()) or from a remote peer on a WebRTC call (using the RTCPeerConnection's audio tracks).

A MediaStreamAudioSourceNode has no inputs and exactly one output, and is created using the AudioContext.createMediaStreamSource() method.

The MediaStreamAudioSourceNode takes the audio from the first MediaStreamTrack whose kind attribute's value is audio. See Track ordering for more information about the order of tracks.

The number of channels output by the node matches the number of tracks found in the selected audio track.

Constructor

new MediaStreamAudioSourceNode()

Creates a new MediaStreamAudioSourceNode object instance with the specified options.

Properties

In addition to the following properties, MediaStreamAudioSourceNode inherits the properties of its parent, AudioNode.

mediaStream Read only

The MediaStream used when constructing this MediaStreamAudioSourceNode.

Methods

Inherits methods from its parent, AudioNode.

Exceptions

InvalidStateError

The stream specified by the mediaStream parameter does not contain any audio tracks.

Usage notes

Typically, you should probably not use this type of node. It has been replaced with the more predictable MediaStreamTrackAudioSourceNode, which has better-defined rules for how it chooses the track to output.

If you do chooose to use MediaStreamAudioSourceNode, you should keep the following in mind.

Track ordering

For the purposes of the MediaStreamTrackAudioSourceNode interface, the order of the audio tracks on the stream is determined by taking the tracks whose kind is audio, then sorting the tracks by their id property's values, in Unicode code point order (essentially, in alphabetical or lexicographical order, for IDs which are simple alphanumeric strings).

The first track, then, is the track whose id comes first when the tracks' IDs are all sorted by Unicode code point.

However, it's important to note that the rule establishing this ordering was added long after this interface was first introduced into the Web Audio API. As such, you can't easily rely on the order matching between any two browsers or browser versions. This is why it is typically wiser to use MediaStreamTrackAudioSourceNode, which provides similar capabilities but was better-defined upon being added to the specification, so it's more reliable.

Example

In this example, we grab a media (audio + video) stream from navigator.getUserMedia, feed the media into a <video> element to play then mute the audio, but then also feed the audio into a MediaStreamAudioSourceNode. Next, we feed this source audio into a low pass BiquadFilterNode (which effectively serves as a bass booster), then a AudioDestinationNode.

The range slider below the <video> element controls the amount of gain given to the lowpass filter — increase the value of the slider to make the audio sound more bass heavy!

Note: You can see this example running live, or view the source.

var pre = document.querySelector('pre');
var video = document.querySelector('video');
var myScript = document.querySelector('script');
var range = document.querySelector('input');

// getUserMedia block - grab stream
// put it into a MediaStreamAudioSourceNode
// also output the visuals into a video element

if (navigator.mediaDevices) {
    console.log('getUserMedia supported.');
    navigator.mediaDevices.getUserMedia ({audio: true, video: true})
    .then(function(stream) {
        video.srcObject = stream;
        video.onloadedmetadata = function(e) {
            video.play();
            video.muted = true;
        };

        // Create a MediaStreamAudioSourceNode
        // Feed the HTMLMediaElement into it
        var audioCtx = new AudioContext();
        var source = audioCtx.createMediaStreamSource(stream);

        // Create a biquadfilter
        var biquadFilter = audioCtx.createBiquadFilter();
        biquadFilter.type = "lowshelf";
        biquadFilter.frequency.value = 1000;
        biquadFilter.gain.value = range.value;

        // connect the AudioBufferSourceNode to the gainNode
        // and the gainNode to the destination, so we can play the
        // music and adjust the volume using the mouse cursor
        source.connect(biquadFilter);
        biquadFilter.connect(audioCtx.destination);

        // Get new mouse pointer coordinates when mouse is moved
        // then set new gain value

        range.oninput = function() {
            biquadFilter.gain.value = range.value;
        }
    })
    .catch(function(err) {
        console.log('The following gUM error occurred: ' + err);
    });
} else {
   console.log('getUserMedia not supported on your browser!');
}

// dump script to pre element

pre.innerHTML = myScript.innerHTML;

Note: As a consequence of calling createMediaStreamSource(), audio playback from the media stream will be re-routed into the processing graph of the AudioContext. So playing/pausing the stream can still be done through the media element API and the player controls.

Specifications

Browser compatibility

BCD tables only load in the browser

See also

• Using the Web Audio API
• WebRTC API
• Media Capture and Streams API (Media Streams)
• MediaStreamTrackAudioSourceNode