Advanced techniques: Creating and sequencing audio - Web APIs
in this tutorial, we're going to cover sound creation and modification, as well as timing and scheduling. we're going to introduce sample loading, envelopes, filters, wavetables, and frequency modulation. if you're familiar with these terms and you're looking for an introduction to their application within with the web audio api, you've come to the right place. demo we're going to be looking at a very simple step sequencer: in practice this is easier to do with a library — the web audio api was built to be built upon. if you are about to embark on building something more complex, tone.js would be a good place to start. however, we want to demonstrate how to build such a demo from first principles, as a learning exercise. note: you can find the source code on github as step-sequence...
...beats in one bar of music. when they are enabled the note will sound. when the instrument plays, it will move across this set of beats and loop the bar. each voice also has local controls, which allow you to manipulate the effects or parameters particular to each technique we are using to create those voices. ...
...te the effects or parameters particular to each technique we are using to create those voices. the techniques we are using are: name of voice technique associated web audio api feature "sweep" oscillator, periodic wave oscillatornode, periodicwave "pulse" multiple oscillators oscillatornode "noise" random noise buffer, biquad filter audiobuffer, audiobuffersourcenode, biquadfilternode "dial up" loading a sound sample to play audiocontext.decodeaudiodata(), audiobuffersourcenode note: this instrument was not created to sound good, it was created to provide demonstration code and represents a very simplified version of such an instrument. the sounds are based on a dial-up modem. if you are unaware of how one s...
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Introducing the Audio API extension - Archive of obsolete content
obsolete since gecko 28 (firefox 28 / thunderbird 28 / seamonkey 2.25 / firefox os 1.3)this feature is obsolete. although it may still work in some browsers, its use is discouraged since it could be removed at any time. try to avoid using it. deprecated since gecko 22 (firefox 22 / thunderbird 22 / seamonkey 2.19)this feature has been removed from the web standards. though some browsers may still support it, it is in the process of being dropped. ...
...pped. avoid using it and update existing code if possible; see the compatibility table at the bottom of this page to guide your decision. be aware that this feature may cease to work at any time. please note that this document describes a non-standard experimental api. this api has been deprecated since gecko 22, disabled since gecko 28, and removed from gecko 31. ...
...ate existing code if possible; see the compatibility table at the bottom of this page to guide your decision. be aware that this feature may cease to work at any time. please note that this document describes a non-standard experimental api. this api has been deprecated since gecko 22, disabled since gecko 28, and removed from gecko 31. you should use the web audio api instead. reading audio streams the loadedmetadata event when the metadata of the media element is available, it triggers a loadedmetadata event. this event has the following attributes: mozchannels: number of channels mozsamplerate: sample rate per second mozframebufferlength: number of samples collected in all channels this information is needed later to decode the audio data stream. the following example ext...
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Video and audio content - Learn web development
previous overview: multimedia and embedding next now that we are comfortable with adding simple images to a webpage, the next step is to start adding video and audio players to your html documents! ...
... previous overview: multimedia and embedding next now that we are comfortable with adding simple images to a webpage, the next step is to start adding video and audio players to your html documents! ...
...ust that with the <video> and <audio> elements; we'll then finish off by looking at how to add captions/subtitles to your videos. prerequisites: basic computer literacy, basic software installed, basic knowledge of working with files, familiarity with html fundamentals (as covered in getting started with html) and images in html. objective: to learn how to embed video and audio content into a webpage, and add captions/subtitles to video. ...
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Using the Web Audio API - Web APIs
let's take a look at getting started with the web audio api. we'll briefly look at some concepts, then study a simple boombox example that allows us to load an audio track, play and pause it, and change its volume and stereo panning. the web audio api does not replace the <audio> media element, but rather complements it, just like <canvas> coexists alongside the <img> element. your use case will determine what tools you use to implement audio. if you simply want to control playback of an audio track, the <audio> media element provides a better, quicker solution than the web audio api. if you want to carry out more complex audio processing, as well as playback, the web audio api provides much more power and control. ...
...for example, there is no ceiling of 32 or 64 sound calls at one time. some processors may be capable of playing more than 1,000 simultaneous sounds without stuttering. ...
...box looks like this: note the retro cassette deck with a play button, and vol and pan sliders to allow you to alter the volume and stereo panning. we could make this a lot more complex, but this is ideal for simple learning at this stage. check out the final demo here on codepen, or see the source code on github. ...
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AudioListener - Web APIs
the audiolistener interface represents the position and orientation of the unique person listening to the audio scene, and is used in audio spatialization. all pannernodes spatialize in relation to the audiolistener stored in the baseaudiocontext.listener attribute. it is important to note that there is only one listener per context and that it isn't an audionode. properties the position, forward, and up value are set and retrieved using different syntaxes. retrieval is done by accessing, for example, audiolistener.positionx, while setting the same property is done with audiolistener.positionx.value. this is why these values are not marked read only, which is how they appear in the specification's idl. ...
...r per context and that it isn't an audionode. properties the position, forward, and up value are set and retrieved using different syntaxes. retrieval is done by accessing, for example, audiolistener.positionx, while setting the same property is done with audiolistener.positionx.value. this is why these values are not marked read only, which is how they appear in the specification's idl. audiolistener.positionx represents the horizontal position of the listener in a right-hand cartesian coordinate sytem. the default is 0. audiolistener.positiony represents the vertical position of the listener in a right-hand cartesian coordinate sytem. the default is 0. ...
...ositionx.value. this is why these values are not marked read only, which is how they appear in the specification's idl. audiolistener.positionx represents the horizontal position of the listener in a right-hand cartesian coordinate sytem. the default is 0. audiolistener.positiony represents the vertical position of the listener in a right-hand cartesian coordinate sytem. the default is 0. audiolistener.positionz represents the longitudinal (back and forth) position of the listener in a right-hand cartesian coordinate sytem. the default is 0. audiolistener.forwardx represents the horizontal position of the listener's forward direction in the same cartesian coordinate sytem as the position (positionx, positiony, and positionz) values. the forward and up values are linearly independent ...
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AudioBufferSourceNode - Web APIs
the audiobuffersourcenode interface is an audioscheduledsourcenode which represents an audio source consisting of in-memory audio data, stored in an audiobuffer. it's especially useful for playing back audio which has particularly stringent timing accuracy requirements, such as for sounds that must match a specific rhythm and can be kept in memory rather than being played from disk or the network. to play sounds which require accurate timing but must be streamed from the network or played from disk, use a audioworkletnode to implement its playback. <div id="interfacediagram" style="display: inline-block; position: relative; width: 100%; padding-bottom: 23.333333333333332%; vertical-align: middle; overflow: hidden;"><svg style="display: inline-block; position: absolute; top: 0; left: 0;" v...
...t="50" fill="#fff" stroke="#d4dde4" stroke-width="2px" /><text x="56" y="30" font-size="12px" font-family="consolas,monaco,andale mono,monospace" fill="#4d4e53" text-anchor="middle" alignment-baseline="middle">eventtarget</text></a><polyline points="111,25 121,20 121,30 111,25" stroke="#d4dde4" fill="none"/><line x1="121" y1="25" x2="151" y2="25" stroke="#d4dde4"/><a xlink:href="/docs/web/api/audionode" target="_top"><rect x="151" y="1" width="90" height="50" fill="#fff" stroke="#d4dde4" stroke-width="2px" /><text x="196" y="30" font-size="12px" font-family="consolas,monaco,andale mono,monospace" fill="#4d4e53" text-anchor="middle" alignment-baseline="middle">audionode</text></a><polyline points="241,25 251,20 251,30 241,25" stroke="#d4dde4" fill="none"/><line x1="251" y1="25" x2="281" ...
...ht="50" fill="#fff" stroke="#d4dde4" stroke-width="2px" /><text x="196" y="30" font-size="12px" font-family="consolas,monaco,andale mono,monospace" fill="#4d4e53" text-anchor="middle" alignment-baseline="middle">audionode</text></a><polyline points="241,25 251,20 251,30 241,25" stroke="#d4dde4" fill="none"/><line x1="251" y1="25" x2="281" y2="25" stroke="#d4dde4"/><a xlink:href="/docs/web/api/audioscheduledsourcenode" target="_top"><rect x="281" y="1" width="240" height="50" fill="#fff" stroke="#d4dde4" stroke-width="2px" /><text x="401" y="30" font-size="12px" font-family="consolas,monaco,andale mono,monospace" fill="#4d4e53" text-anchor="middle" alignment-baseline="middle">audioscheduledsourcenode</text></a><polyline points="521,25 531,20 531,30 521,25" stroke="#d4dde4" fill="none"/><...
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Basic concepts behind Web Audio API - Web APIs
this article explains some of the audio theory behind how the features of the web audio api work, to help you make informed decisions while designing how audio is routed through your app. it won't turn you into a master sound engineer, but it will give you enough background to understand why the web audio api works like it does. audio graphs the web audio api involves handling audio operations inside an audio context, and has been designed to allow modular routing. basic audio operations are performed with audio nodes, which are linked together to form an audio routing graph. several sources — with different types of channel layout — are supported even within a single context. this modular design provides the flexibility to create complex audio functions with dynamic effects. a...
...ported even within a single context. this modular design provides the flexibility to create complex audio functions with dynamic effects. audio nodes are linked via their inputs and outputs, forming a chain that starts with one or more sources, goes through one or more nodes, then ends up at a destination. although, you don't have to provide a destination if you, say, just want to visualize some audio data. a simple, typical workflow for web audio would look something like this: create the audio context. inside the context, create sources — such as <audio>, oscillator, or stream. create effects nodes, such as reverb, biquad filter, panner, or compressor. ...
...although, you don't have to provide a destination if you, say, just want to visualize some audio data. a simple, typical workflow for web audio would look something like this: create the audio context. inside the context, create sources — such as <audio>, oscillator, or stream. create effects nodes, such as reverb, biquad filter, panner, or compressor. choose the final destination for the audio, such as the user's computer speakers. establish connections from the audio sources through zero or more effects, eventually ending at the chosen destination. channel notation the number of audio channels available on a signal is frequently presented in a numeric format, such as 2.0 or 5.1. ...
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BaseAudioContext.createBuffer() - Web APIs
the createbuffer() method of the baseaudiocontext interface is used to create a new, empty audiobuffer object, which can then be populated by data, and played via an audiobuffersourcenode for more details about audio buffers, check out the audiobuffer reference page. note: createbuffer() used to be able to take compressed data and give back decoded samples, but this ability was removed from the spec, because all the decoding was done on the main thread, therefore createbuffer() was blocking other code execution. the asynchronous method decodeaudiodata() does the same thing — takes compressed audio, say, an mp3 file, and directly gives you back an audiobuffer that you can then set to play via in an audiobuffersourcenode. ...
...y was removed from the spec, because all the decoding was done on the main thread, therefore createbuffer() was blocking other code execution. the asynchronous method decodeaudiodata() does the same thing — takes compressed audio, say, an mp3 file, and directly gives you back an audiobuffer that you can then set to play via in an audiobuffersourcenode. for simple uses like playing an mp3, decodeaudiodata() is what you should be using. syntax var buffer = baseaudiocontext.createbuffer(numofchannels, length, samplerate); parameters note: for an in-depth explanation of how audio buffers work, and what these parameters mean, read audio buffers: frames, samples and channels from our basic concepts guide. numofchannels an integer representing the number of channels this buffer should hav...
.... numofchannels an integer representing the number of channels this buffer should have. the default value is 1, and all user agents must support at least 32 channels. length an integer representing the size of the buffer in sample-frames (where each sample-frame is the size of a sample in bytes multiplied by numofchannels). to determine the length to use for a specific number of seconds of audio, use numseconds * samplerate. samplerate the sample rate of the linear audio data in sample-frames per second. all browsers must support sample rates in at least the range 8,000 hz to 96,000 hz. returns an audiobuffer configured based on the specified options. exceptions notsupportederror one or more of the options are negative or otherwise has an invalid value (such as numberofchannels...
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BaseAudioContext - Web APIs
the baseaudiocontext interface of the web audio api acts as a base definition for online and offline audio-processing graphs, as represented by audiocontext and offlineaudiocontext respectively. you wouldn't use baseaudiocontext directly — you'd use its features via one of these two inheriting interfaces. a baseaudiocontext can be a target of events, therefore it implements the eventtarget interface. <div id="interfacediagram" style="display: inline-block; position: relative; width: 100%; padding-bottom: 11.666666666666666%; vertical-align: middle; overflow: hidden;"><svg style="display: inline-block; position: absolute; top: 0; left: 0;" viewbox="-50 0 600 70" preserveaspectratio="xminymin meet"><a xlink:href="/docs/web/api/eventtarget" target="_top"><rect x="1" y="1" width="110" heig...
...0" fill="#fff" stroke="#d4dde4" stroke-width="2px" /><text x="56" y="30" font-size="12px" font-family="consolas,monaco,andale mono,monospace" fill="#4d4e53" text-anchor="middle" alignment-baseline="middle">eventtarget</text></a><polyline points="111,25 121,20 121,30 111,25" stroke="#d4dde4" fill="none"/><line x1="121" y1="25" x2="151" y2="25" stroke="#d4dde4"/><a xlink:href="/docs/web/api/baseaudiocontext" target="_top"><rect x="151" y="1" width="160" height="50" fill="#f4f7f8" stroke="#d4dde4" stroke-width="2px" /><text x="231" y="30" font-size="12px" font-family="consolas,monaco,andale mono,monospace" fill="#4d4e53" text-anchor="middle" alignment-baseline="middle">baseaudiocontext</text></a></svg></div> a:hover text { fill: #0095dd; pointer-events: all;} properties baseaudio...
...it starts at 0. bas...
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Audio and video manipulation - Developer guides
this article provides a reference to explain what you need to do. video manipulation the ability to read the pixel values from each frame of a video can be very useful. video and canvas the <canvas> element provides a surface for drawing graphics onto web pages; it is very powerful and can be coupled tightly with video. ...
...n resource sharing) on your video server. video and webgl webgl is a powerful api that uses canvas to draw hardware-accelerated 3d or 2d scenes. you can combine webgl and the <video> element to create video textures, which means you can put video inside 3d scenes. note: you can find the source code of this demo on github (see it live also). playback rate we can also adjust the rate that audio and video plays at using an attribute of the <audio> and <video> element called playbackrate. playbackrate is a number that represents a multiple to be applied to the rate of playback, for example 0.5 represents half speed while 2 represents double speed. note that the playbackrate property works with both <audio> and <video>, but in both cases, it changes the playback speed but not the pitch. t...
...at using an attribute of the <audio> and <video> element called playbackrate. playbackrate is a number that represents a multiple to be applied to the rate of playback, for example 0.5 represents half speed while 2 represents double speed. note that the playbackrate property works with both <audio> and <video>, but in both cases, it changes the playback speed but not the pitch. to manipulate the audio's pitch you need to use the web audio api. see the audiobuffersourcenode.playbackrate property. html <video id="my-video" controls src="https://udn.realityripple.com/samples/6f/08625b424a.m4v"> </video> javascript var myvideo = document.getelementbyid('my-video'); myvideo.playbackrate = 2; playable code <video id="my-video" controls="true" width="480" height="270"> <source src="ht...
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AudioNode.connect() - Web APIs
the connect() method of the audionode interface lets you connect one of the node's outputs to a target, which may be either another audionode (thereby directing the sound data to the specified node) or an audioparam, so that the node's output data is automatically used to change the value of that parameter over time. syntax var destinationnode = audionode.connect(destination, outputindex, inputindex); audionode.connect(destination, outputindex); parameters destination the audionode or audioparam to which to connect. outputindex optional an index specifying which output of the current audionode to connect to the destination. the index numbers are defined according to the number of output channels (see audio channels). while you can only connect a given output to a given input once ...
...this is allowed only if there is at least one delaynode in the cycle. ...
...to a given input once (repeated attempts are ignored), you can connect an output to multiple inputs by calling connect() repeatedly. this makes fan-out possible. the default value is 0. inputindex optional an index describing which input of the destination you want to connect the current audionode to; the default is 0. the index numbers are defined according to the number of input channels (see audio channels). it is possible to connect an audionode to another audionode, which in turn connects back to the first audionode, creating a cycle. this is allowed only if there is at least one delaynode in the cycle. otherwise, a notsupportederror exception is thrown. this parameter is not allowed if the destination is an audioparam. return value if the destination is a node, connect() returns a re...
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AudioWorkletNode.parameters - Web APIs
there we've created a simple node which outputs white noise. ...
...there we've created a simple node which outputs white noise. ...
...there we've created a simple node which outputs white noise. ...
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OfflineAudioContext - Web APIs
the offlineaudiocontext interface is an audiocontext interface representing an audio-processing graph built from linked together audionodes. in contrast with a standard audiocontext, an offlineaudiocontext doesn't render the audio to the device hardware; instead, it generates it, as fast as it can, and outputs the result to an audiobuffer. <div id="interfacediagram" style="display: inline-block; position: relative; width: 100%; padding-bottom: 11.666666666666666%; vertical-align: middle; overflow: hidden;"><svg style="display: inline-block; position: absolute; top: 0; left: 0;" viewbox="-50 0 600 70" preserveaspectratio="xminymin meet"><a xlink:href="/docs/web/api/eventtarget" target="_top"><rect x="1" y="1" width="110" height="50" fill="#fff" stroke="#d4dde4" stroke-width="2px" /><text ...
...t="50" fill="#fff" stroke="#d4dde4" stroke-width="2px" /><text x="56" y="30" font-size="12px" font-family="consolas,monaco,andale mono,monospace" fill="#4d4e53" text-anchor="middle" alignment-baseline="middle">eventtarget</text></a><polyline points="111,25 121,20 121,30 111,25" stroke="#d4dde4" fill="none"/><line x1="121" y1="25" x2="151" y2="25" stroke="#d4dde4"/><a xlink:href="/docs/web/api/audiocontext" target="_top"><rect x="151" y="1" width="120" height="50" fill="#fff" stroke="#d4dde4" stroke-width="2px" /><text x="211" y="30" font-size="12px" font-family="consolas,monaco,andale mono,monospace" fill="#4d4e53" text-anchor="middle" alignment-baseline="middle">audiocontext</text></a><polyline points="271,25 281,20 281,30 271,25" stroke="#d4dde4" fill="none"/><line x1="281" y1="25" x2...
...ll="#fff" stroke="#d4dde4" stroke-width="2px" /><text x="211" y="30" font-size="12px" font-family="consolas,monaco,andale mono,monospace" fill="#4d4e53" text-anchor="middle" alignment-baseline="middle">audiocontext</text></a><polyline points="271,25 281,20 281,30 271,25" stroke="#d4dde4" fill="none"/><line x1="281" y1="25" x2="311" y2="25" stroke="#d4dde4"/><a xlink:href="/docs/web/api/offlineaudiocontext" target="_top"><rect x="311" y="1" width="190" height="50" fill="#f4f7f8" stroke="#d4dde4" stroke-width="2px" /><text x="406" y="30" font-size="12px" font-family="consolas,monaco,andale mono,monospace" fill="#4d4e53" text-anchor="middle" alignment-baseline="middle">offlineaudiocontext</text></a></svg></div> a:hover text { fill: #0095dd; pointer-events: all;} constructor offli...
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BaseAudioContext.createScriptProcessor() - Web APIs
if specified, the buffersize must be one of the following values: 256, 512, 1024, 2048, 4096, 8192, 16384. ...
...if specified, the buffersize must be one of the following values: 256, 512, 1024, 2048, 4096, 8192, 16384. ...
...if specified, the buffersize must be one of the following values: 256, 512, 1024, 2048, 4096, 8192, 16384. ...
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AudioNode - Web APIs
the audionode interface is a generic interface for representing an audio processing module. examples include: an audio source (e.g. an html <audio> or <video> element, an oscillatornode, etc.), the audio destination, intermediate processing module (e.g. a filter like biquadfilternode or convolvernode), or volume control (like gainnode) <div id="interfacediagram" style="display: inline-block; position: relative; width: 100%; padding-bottom: 11.666666666666666%; vertical-align: middle; overflow: hidden;"><svg style="display: inline-block; position: absolute; top: 0; left: 0;" viewbox="-50 0 600 70" preserveaspectratio="xminymin meet"><a xlink:href="/docs/web/api/eventtarget" target="_top"><rect x="1" y="1" width="110" height="50" fill="#fff" stroke="#d4dde4" stroke-width="2px" /><te...
...t="50" fill="#fff" stroke="#d4dde4" stroke-width="2px" /><text x="56" y="30" font-size="12px" font-family="consolas,monaco,andale mono,monospace" fill="#4d4e53" text-anchor="middle" alignment-baseline="middle">eventtarget</text></a><polyline points="111,25 121,20 121,30 111,25" stroke="#d4dde4" fill="none"/><line x1="121" y1="25" x2="151" y2="25" stroke="#d4dde4"/><a xlink:href="/docs/web/api/audionode" target="_top"><rect x="151" y="1" width="90" height="50" fill="#f4f7f8" stroke="#d4dde4" stroke-width="2px" /><text x="196" y="30" font-size="12px" font-family="consolas,monaco,andale mono,monospace" fill="#4d4e53" text-anchor="middle" alignment-baseline="middle">audionode</text></a></svg></div> a:hover text { fill: #0095dd; pointer-events: all;} note: an audionode can be target o...
...troke="#d4dde4" stroke-width="2px" /><text x="196" y="30" font-size="12px" font-family="consolas,monaco,andale mono,monospace" fill="#4d4e53" text-anchor="middle" alignment-baseline="middle">audionode</text></a></svg></div> a:hover text { fill: #0095dd; pointer-events: all;} note: an audionode can be target of events, therefore it implements the eventtarget interface. description the audio routing graph each audionode has inputs and outputs, and multiple audio nodes are connected to build a processing graph. this graph is contained in an audiocontext, and each audio node can only belong to one audio context. a source node has zero inputs but one or multiple outputs, and can be used to generate sound. on the other hand, a destination node has no outputs; instead, all its inputs ...
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AudioParam.setValueAtTime() - Web APIs
in some browsers older implementations of this interface return void. ...
...in some browsers older implementations of this interface return void. ...
...in some browsers older implementations of this interface return void. ...
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Web Audio API best practices - Web APIs
there's no strict right or wrong way when writing creative code. as long as you consider security, performance, and accessibility, you can adapt to your own style. in this article, we'll share a number of best practices — guidelines, tips, and tricks for working with the web audio api. loading sounds/files there are four main ways to load sound with the web audio api and it can be a little confusing as to which one you should use. when working with files, you are looking at either the grabbing the file from an htmlmediaelement (i.e. an <audio> or <video> element), or you're looking to fetch the file and decode it into a buffer. both are legitimate ways of working, however, it's more common to use the former when you are working with full-length tracks, and the latter when working with sho...
... you will, however, have more control if you use a buffer node. ...
... ways of working, however, it's more common to use the former when you are working with full-length tracks, and the latter when working with shorter, more sample-like tracks. media elements have streaming support out of the box. the audio will start playing when the browser determines it can load the rest of the file before playing finishes. you can see an example of how to use this with the web audio api in the using the web audio api tutorial. you will, however, have more control if you use a buffer node. ...
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Autoplay guide for media and Web Audio APIs - Web media technologies
automatically starting the playback of audio (or videos with audio tracks) immediately upon page load can be an unwelcome surprise to users. while autoplay of media serves a useful purpose, it should be used carefully and only when needed. in order to give users control over this, browsers often provide various forms of autoplay blocking. in this guide, we'll cover autoplay functionality in the various media and web audio apis, including a brief overview of how to use autoplay and how to work with browsers to handle autoplay blocking gracefully. autoplay blocking is not applied to <video> elements when the source media does not have an audio track, or if the audio track is muted. media with an active audio track are considered to be audible, and autoplay blocking applies to them. inaud...
...locking gracefully. autoplay blocking is not applied to <video> elements when the source media does not have an audio track, or if the audio track is muted. media with an active audio track are considered to be audible, and autoplay blocking applies to them. inaudible media are not affected by autoplay blocking. autoplay and autoplay blocking the term autoplay refers to any feature that causes audio to begin to play without the user specifically requesting that playback begin. this includes both the use of html attributes to autoplay media as well as the user of javascript code to start playback outside the context of handling user input. ...
...auses audio to begin to play without the user specifically requesting that playback begin. this includes both the use of html attributes to autoplay media as well as the user of javascript code to start playback outside the context of handling user input. that means that both of the following are considered autoplay behavior, and are therefore subject to the browser's autoplay blocking policy: <audio src="/music.mp4" autoplay> and audioelement.play(); the following web features and apis may be affected by autoplay blocking: the html <audio> and <video> elements the web audio api from the user's perspective, a web page or app that spontaneously starts making noise without warning can be jarring, inconvenient, or off-putting. because of that, browsers generally only allow autoplay to o...
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AudioBuffer - Web APIs
the audiobuffer interface represents a short audio asset residing in memory, created from an audio file using the audiocontext.decodeaudiodata() method, or from raw data using audiocontext.createbuffer(). once put into an audiobuffer, the audio can then be played by being passed into an audiobuffersourcenode. objects of these types are designed to hold small audio snippets, typically less than 45 s. for longer sounds, objects implementing the mediaelementaudiosourcenode are more suitable. the buffer contains data in the following format: non-interleaved ieee754 32-bit linear pcm with a nominal range between -1 and +1, that is, 32bits floating point buffer, with each samples between -1.0 and 1.0. if the audiobuffer has multiple channels, they are stored in separate buffer. constructor audi...
...ng the mediaelementaudiosourcenode are more suitable. the buffer contains data in the following format: non-interleaved ieee754 32-bit linear pcm with a nominal range between -1 and +1, that is, 32bits floating point buffer, with each samples between -1.0 and 1.0. if the audiobuffer has multiple channels, they are stored in separate buffer. constructor audiobuffer() creates and returns a new audiobuffer object instance. properties audiobuffer.samplerate read only returns a float representing the sample rate, in samples per second, of the pcm data stored in the buffer. audiobuffer.length read only returns an integer representing the length, in sample-frames, of the pcm data stored in the buffer. ...
...ed in separate buffer. constructor audiobuffer() creates and returns a new audiobuffer object instance. properties audiobuffer.samplerate read only returns a float representing the sample rate, in samples per second, of the pcm data stored in the buffer. audiobuffer.length read only returns an integer representing the length, in sample-frames, of the pcm data stored in the buffer. audiobuffer.duration read only returns a double representing the duration, in seconds, of the pcm data stored in the buffer. audiobuffer.numberofchannels read only returns an integer representing the number of discrete audio channels described by the pcm data stored in the buffer. ...
...And 14 more matches

AudioListener.setOrientation() - Web APIs
the setorientation() method of the audiolistener interface defines the orientation of the listener. it consists of two direction vectors: the front vector, defined by the three unitless parameters x, y and z, describes the direction of the face of the listener, that is the direction the nose of the person is pointing towards. the front vector's default value is (0, 0, -1). the up vector, defined by three unitless parameters xup, yup and zup, describes the direction of the top of the listener's head. the up vector's default value is (0, 1, 0). the two vectors must be separated by an angle of 90° — in linear analysis terms, they must be perpendicular to each other. ...
... of the person is pointing towards. the front vector's default value is (0, 0, -1). the up vector, defined by three unitless parameters xup, yup and zup, describes the direction of the top of the listener's head. the up vector's default value is (0, 1, 0). the two vectors must be separated by an angle of 90° — in linear analysis terms, they must be perpendicular to each other. ...
...is pointing towards. the front vector's default value is (0, 0, -1). the up vector, defined by three unitless parameters xup, yup and zup, describes the direction of the top of the listener's head. the up vector's default value is (0, 1, 0). the two vectors must be separated by an angle of 90° — in linear analysis terms, they must be perpendicular to each other. ...
...And 14 more matches

AudioParamDescriptor - Web APIs
the audioparamdescriptor dictionary of the web audio api specifies properties for an audioparam objects. it is used to create custom audioparams on an audioworkletnode. if the underlying audioworkletprocessor has a parameterdescriptors static getter, then the returned array of objects based on this dictionary is used internally by audioworkletnode constructor to populate its parameters property accordingly. properties name the domstring which represents the name of the audioparam. under this name the audioparam will be available in the parameters property of the node, and under this name the audioworkletprocessor.process method will acquire the calculated values of this audioparam. automationrate optional either "a-rate", or "k-rate" string which represents an automation rate of this au...
...name the audioparam will be available in the parameters property of the node, and under this name the audioworkletprocessor.process method will acquire the calculated values of this audioparam. automationrate optional either "a-rate", or "k-rate" string which represents an automation rate of this audioparam. defaults to "a-rate". minvalue optional a float which represents minimum value of the audioparam. defaults to -3.4028235e38. maxvalue optional a float which represents maximum value of the audioparam. defaults to 3.4028235e38. defaultvalue optional a float which represents initial value of the audioparam. defaults to 0. examples to demonstrate creation and usage of custom audioparams, we'll expand the example from audioworkletnode page. there we've created a simple node which out...
...m. defaults to 0. examples to demonstrate creation and usage of custom audioparams, we'll expand the example from audioworkletnode page. there we've created a simple node which outputs white noise. here, additionally, we'll create a custom gain parameter, so we can directly change volume of the output (although you could use gainnode to achieve this as well). first, we need to define a custom audioworkletprocessor, and register it. note that this should be done in a separate file. we expand the processor by adding a static parameterdescriptors getter. it will be used internally by the audioworkletnode constructor to populate its parameters with instantiated audioparam objects. ...
...And 14 more matches

AudioWorkletProcessor() - Web APIs
available properties are as follows: numberofinputs optional the value to initialize the numberofinputs property to. defaults to 1. ...
...available properties are as follows: numberofinputs optional the value to initialize the numberofinputs property to. defaults to 1. ...
...available properties are as follows: numberofinputs optional the value to initialize the numberofinputs property to. defaults to 1. ...
...And 14 more matches

AudioWorkletProcessor.parameterDescriptors (static getter) - Web APIs
the read-only parameterdescriptors property of an audioworkletprocessor-derived class is a static getter, which returns an iterable of audioparamdescriptor-based objects. the property is not a part of the audioworkletprocessor interface, but, if defined, it is called internally by the audioworkletprocessor constructor to create a list of custom audioparam objects in the parameters property of the associated audioworkletnode. defining the getter is optional. syntax audioworkletprocessorsubclass.parameterdescriptors; value an iterable of audioparamdescriptor-based objects. the properties of these objects are as follows: name the domstring which represents the name of the audioparam. ...
...r to create a list of custom audioparam objects in the parameters property of the associated audioworkletnode. defining the getter is optional. syntax audioworkletprocessorsubclass.parameterdescriptors; value an iterable of audioparamdescriptor-based objects. the properties of these objects are as follows: name the domstring which represents the name of the audioparam. under this name the audioparam will be available in the parameters property of the node, and under this name the audioworkletprocessor.process method will acquire the calculated values of this audioparam. automationrate optional either "a-rate", or "k-rate" string which represents an automation rate of this audioparam. defaults to "a-rate". minvalue optional a float which represents minimum value of the audioparam. de...
...there we've created a simple node which outputs white noise. ...
...And 14 more matches

MediaElementAudioSourceNode - Web APIs
the mediaelementaudiosourcenode interface represents an audio source consisting of an html5 <audio> or <video> element. it is an audionode that acts as an audio source. a mediaelementsourcenode has no inputs and exactly one output, and is created using the audiocontext.createmediaelementsource() method. the amount of channels in the output equals the number of channels of the audio referenced by the htmlmediaelement used in the creation of the node, or is 1 if the htmlmediaelement has no audio. number of inputs 0 number of outputs 1 channel count defined by the media in the htmlmediaelement passed to the audiocontext.createmediaelementsource method that created it. ...
...als the number of channels of the audio referenced by the htmlmediaelement used in the creation of the node, or is 1 if the htmlmediaelement has no audio. number of inputs 0 number of outputs 1 channel count defined by the media in the htmlmediaelement passed to the audiocontext.createmediaelementsource method that created it. constructor mediaelementaudiosourcenode() creates a new mediaelementaudiosourcenode object instance. properties inherits properties from its parent, audionode. mediaelement read only the htmlmediaelement used when constructing this mediastreamaudiosourcenode. methods inherits methods from its parent, audionode. ...
...when the mouse pointer is moved, the updatepage() function is invoked, which calculates the current gain as a ratio of mouse y position divided by overall window height. ...
...And 14 more matches

AudioContext.createMediaElementSource() - Web APIs
the createmediaelementsource() method of the audiocontext interface is used to create a new mediaelementaudiosourcenode object, given an existing html <audio> or <video> element, the audio from which can then be played and manipulated. for more details about media element audio source nodes, check out the mediaelementaudiosourcenode reference page. syntax var audioctx = new audiocontext(); var source = audioctx.createmediaelementsource(mymediaelement); parameters mymediaelement an htmlmediaelement object that you want to feed into an audio processing graph to manipulate. returns a mediaelementaudiosourcenode. ...
...when the mouse pointer is moved, the updatepage() function is invoked, which calculates the current gain as a ratio of mouse y position divided by overall window height. ...
...when the mouse pointer is moved, the updatepage() function is invoked, which calculates the current gain as a ratio of mouse y position divided by overall window height. you can therefore increase and decrease the volume of the playing music by moving the mouse pointer up and down. ...
...And 13 more matches

AudioProcessingEvent - Web APIs
properties the list below includes the properties inherited from its parent, event. property type description target read only eventtarget the event target (the topmost target in the dom tree). type read only domstring the type of event. ...
...e description target read only eventtarget the event target (the topmost target in the dom tree). type read only domstring the type of event. ...
... eventtarget the event target (the topmost target in the dom tree). type read only domstring the type of event. ...
...And 12 more matches

MediaStreamAudioSourceNode - Web APIs
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. number of...
... 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. number of inputs 0 number of outputs 1 channel count defined by the first audio mediastreamtrack passed to the audiocontext.createmediastreamsource() method that created it. 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 m...
...emediastreamsource() method that created it. 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. ...
...And 12 more matches

MediaStreamConstraints.audio - Web APIs
the mediastreamconstraints dictionary's audio property is used to indicate what kind of audio track, if any, should be included in the mediastream returned by a call to getusermedia(). to learn more about how constraints work, see capabilities, constraints, and settings. syntax var audioconstraints = true | false | mediatrackconstraints; value the value of the audio property can be specified as either of two types: boolean if a boolean value is specified, it simply indicates whether or not an audio track should be included in the returned stream; if it's true, an audio track is included; if no audio source is available or if permission is not given to use the audio source, the call to getusermedia() will fail. if false, no audio track is included. mediatrackconstraints a constra...
...eam; if it's true, an audio track is included; if no audio source is available or if permission is not given to use the audio source, the call to getusermedia() will fail. if false, no audio track is included. mediatrackconstraints a constraints dictionary detailing the preferable and/or required values or ranges of values for the track's constrainable properties. if you specify constraints, an audio track meeting the constraints is required. examples for browsers with feature policy enabled, the samples below need the microphone feature enabled. see security in mediadevices.getusermedia() for details and examples on how to configure this. using a boolean value in this example, we provide a simple value of true for the audio property. ...
...an audio track meeting the constraints is required. examples for browsers with feature policy enabled, the samples below need the microphone feature enabled. see security in mediadevices.getusermedia() for details and examples on how to configure this. using a boolean value in this example, we provide a simple value of true for the audio property. this tells getusermedia() that we require an audio track, but we don't care about any specifics beyond that. ...
...And 12 more matches

OfflineAudioContext.startRendering() - Web APIs
the startrendering() method of the offlineaudiocontext interface starts rendering the audio graph, taking into account the current connections and the current scheduled changes. the complete event (of type offlineaudiocompletionevent) is raised when the rendering is finished, containing the resulting audiobuffer in its renderedbuffer property. browsers currently support two versions of the startrendering() method — an older event-based version and a newer promise-based version. the former will eventually be removed, but currently both mechanisms are provided for legacy reasons. ...
...event) is raised when the rendering is finished, containing the resulting audiobuffer in its renderedbuffer property. browsers currently support two versions of the startrendering() method — an older event-based version and a newer promise-based version. the former will eventually be removed, but currently both mechanisms are provided for legacy reasons. syntax event-based version: offlineaudioctx.startrendering(); offlineaudioctx.oncomplete = function(e) { // e.renderedbuffer contains the output buffer } promise-based version: offlineaudioctx.startrendering().then(function(buffer) { // buffer contains the output buffer }); parameters none. returns void. ...
...vided for legacy reasons. syntax event-based version: offlineaudioctx.startrendering(); offlineaudioctx.oncomplete = function(e) { // e.renderedbuffer contains the output buffer } promise-based version: offlineaudioctx.startrendering().then(function(buffer) { // buffer contains the output buffer }); parameters none. returns void. example in this simple example, we declare both an audiocontext and an offlineaudiocontext object. we use the audiocontext to load an audio track via xhr (audiocontext.decodeaudiodata), then the offlineaudiocontext to render the audio into an audiobuffersourcenode and play the track through. after the offline audio graph is set up, you need to render it to an audiobuffer using offlineaudiocontext.startrendering. when the startrendering() promise resol...
...And 12 more matches

AudioListener.forwardX - Web APIs
the forwardx read-only property of the audiolistener interface is an audioparam representing the x value of the direction vector defining the forward direction the listener is pointing in. note: the parameter is a-rate when used with a pannernode whose panningmodel is set to equalpower, or k-rate otherwise. syntax var audioctx = new audiocontext(); var mylistener = audioctx.listener; mylistener.forwardx.value = 0; value an audioparam. its default value is 0, and it can range between positive and negative infinity. example in the following example, you can see an example of how the createpanner() method, audiolistener and pannernode would be used to control audio spatialisation. ...
... mylistener = audioctx.listener; mylistener.forwardx.value = 0; value an audioparam. its default value is 0, and it can range between positive and negative infinity. example in the following example, you can see an example of how the createpanner() method, audiolistener and pannernode would be used to control audio spatialisation. generally you will define the position in 3d space that your audio listener and panner (source) occupy initially, and then update the position of one or both of these as the application is used. you might be moving a character around inside a game world for example, and wanting delivery of audio to change realistically as your character moves closer to or further away from a music player such as a stereo. in the example you can see this being controlled by the f...
...tion via the positionpanner() function. to see a complete implementation, check out our panner-node example (view the source code) — this demo transports you to the 2.5d "room of metal", where you can play a track on a boom box and then walk around the boom box to see how the sound changes! note how we have used some feature detection to either give the browser the newer property values (like audiolistener.forwardx) for setting position, etc. if it supports those, or older methods (like audiolistener.setorientation()) if it still supports those but not the new properties. // set up listener and panner position information var width = window.innerwidth; var height = window.innerheight; var xpos = math.floor(width/2); var ypos = math.floor(height/2); var zpos = 295; // define other variabl...
...And 11 more matches

AudioListener.forwardY - Web APIs
the forwardy read-only property of the audiolistener interface is an audioparam representing the y value of the direction vector defining the forward direction the listener is pointing in. note: the parameter is a-rate when used with a pannernode whose panningmodel is set to equalpower, or k-rate otherwise. syntax var audioctx = new audiocontext(); var mylistener = audioctx.listener; mylistener.forwardy.value = 0; value an audioparam. its default value is 0, and it can range between positive and negative infinity. example in the following example, you can see an example of how the createpanner() method, audiolistener and pannernode would be used to control audio spatialisation. ...
... mylistener = audioctx.listener; mylistener.forwardy.value = 0; value an audioparam. its default value is 0, and it can range between positive and negative infinity. example in the following example, you can see an example of how the createpanner() method, audiolistener and pannernode would be used to control audio spatialisation. generally you will define the position in 3d space that your audio listener and panner (source) occupy initially, and then update the position of one or both of these as the application is used. you might be moving a character around inside a game world for example, and wanting delivery of audio to change realistically as your character moves closer to or further away from a music player such as a stereo. in the example you can see this being controlled by the f...
...tion via the positionpanner() function. to see a complete implementation, check out our panner-node example (view the source code) — this demo transports you to the 2.5d "room of metal", where you can play a track on a boom box and then walk around the boom box to see how the sound changes! note how we have used some feature detection to either give the browser the newer property values (like audiolistener.forwardx) for setting position, etc. if it supports those, or older methods (like audiolistener.setorientation()) if it still supports those but not the new properties. // set up listener and panner position information var width = window.innerwidth; var height = window.innerheight; var xpos = math.floor(width/2); var ypos = math.floor(height/2); var zpos = 295; // define other variabl...
...And 11 more matches

AudioListener.forwardZ - Web APIs
the forwardz read-only property of the audiolistener interface is an audioparam representing the z value of the direction vector defining the forward direction the listener is pointing in. note: the parameter is a-rate when used with a pannernode whose panningmodel is set to equalpower, or k-rate otherwise. syntax var audioctx = new audiocontext(); var mylistener = audioctx.listener; mylistener.forwardz.value = 0; value an audioparam. its default value is -1, and it can range between positive and negative infinity. example in the following example, you can see an example of how the createpanner() method, audiolistener and pannernode would be used to control audio spatialisation. ...
...mylistener = audioctx.listener; mylistener.forwardz.value = 0; value an audioparam. its default value is -1, and it can range between positive and negative infinity. example in the following example, you can see an example of how the createpanner() method, audiolistener and pannernode would be used to control audio spatialisation. generally you will define the position in 3d space that your audio listener and panner (source) occupy initially, and then update the position of one or both of these as the application is used. you might be moving a character around inside a game world for example, and wanting delivery of audio to change realistically as your character moves closer to or further away from a music player such as a stereo. in the example you can see this being controlled by the f...
...tion via the positionpanner() function. to see a complete implementation, check out our panner-node example (view the source code) — this demo transports you to the 2.5d "room of metal", where you can play a track on a boom box and then walk around the boom box to see how the sound changes! note how we have used some feature detection to either give the browser the newer property values (like audiolistener.forwardx) for setting position, etc. if it supports those, or older methods (like audiolistener.setorientation()) if it still supports those but not the new properties. // set up listener and panner position information var width = window.innerwidth; var height = window.innerheight; var xpos = math.floor(width/2); var ypos = math.floor(height/2); var zpos = 295; // define other variabl...
...And 11 more matches

AudioListener.positionX - Web APIs
the positionx read-only property of the audiolistener interface is an audioparam representing the x position of the listener in 3d cartesian space. note: the parameter is a-rate when used with a pannernode whose pannernode is set to equalpower, or k-rate otherwise. syntax var audioctx = new audiocontext(); var mylistener = audioctx.listener; mylistener.positionx.value = 1; value an audioparam. its default value is 0, and it can range between positive and negative infinity. example in the following example, you can see an example of how the createpanner() method, audiolistener and pannernode would be used to control audio spatialisation. ...
...mylistener = audioctx.listener; mylistener.positionx.value = 1; value an audioparam. its default value is 0, and it can range between positive and negative infinity. example in the following example, you can see an example of how the createpanner() method, audiolistener and pannernode would be used to control audio spatialisation. generally you will define the position in 3d space that your audio listener and panner (source) occupy initially, and then update the position of one or both of these as the application is used. you might be moving a character around inside a game world for example, and wanting delivery of audio to change realistically as your character moves closer to or further away from a music player such as a stereo. in the example you can see this being controlled by the f...
...tion via the positionpanner() function. to see a complete implementation, check out our panner-node example (view the source code) — this demo transports you to the 2.5d "room of metal", where you can play a track on a boom box and then walk around the boom box to see how the sound changes! note how we have used some feature detection to either give the browser the newer property values (like audiolistener.forwardx) for setting position, etc. if it supports those, or older methods (like audiolistener.setorientation()) if it still supports those but not the new properties. // set up listener and panner position information var width = window.innerwidth; var height = window.innerheight; var xpos = math.floor(width/2); var ypos = math.floor(height/2); var zpos = 295; // define other variabl...
...And 11 more matches

AudioListener.positionY - Web APIs
the positiony read-only property of the audiolistener interface is an audioparam representing the y position of the listener in 3d cartesian space. note: the parameter is a-rate when used with a pannernode whose pannernode is set to equalpower, or k-rate otherwise. syntax var audioctx = new audiocontext(); var mylistener = audioctx.listener; mylistener.positiony.value = 1; value an audioparam. its default value is 0, and it can range between positive and negative infinity. example in the following example, you can see an example of how the createpanner() method, audiolistener and pannernode would be used to control audio spatialisation. ...
...mylistener = audioctx.listener; mylistener.positiony.value = 1; value an audioparam. its default value is 0, and it can range between positive and negative infinity. example in the following example, you can see an example of how the createpanner() method, audiolistener and pannernode would be used to control audio spatialisation. generally you will define the position in 3d space that your audio listener and panner (source) occupy initially, and then update the position of one or both of these as the application is used. you might be moving a character around inside a game world for example, and wanting delivery of audio to change realistically as your character moves closer to or further away from a music player such as a stereo. in the example you can see this being controlled by the f...
...tion via the positionpanner() function. to see a complete implementation, check out our panner-node example (view the source code) — this demo transports you to the 2.5d "room of metal", where you can play a track on a boom box and then walk around the boom box to see how the sound changes! note how we have used some feature detection to either give the browser the newer property values (like audiolistener.forwardx) for setting position, etc. if it supports those, or older methods (like audiolistener.setorientation()) if it still supports those but not the new properties. // set up listener and panner position information var width = window.innerwidth; var height = window.innerheight; var xpos = math.floor(width/2); var ypos = math.floor(height/2); var zpos = 295; // define other variabl...
...And 11 more matches

AudioListener.positionZ - Web APIs
the positionz read-only property of the audiolistener interface is an audioparam representing the z position of the listener in 3d cartesian space. note: the parameter is a-rate when used with a pannernode whose pannernode is set to equalpower, or k-rate otherwise. syntax var audioctx = new audiocontext(); var mylistener = audioctx.listener; mylistener.positionz.value = 1; value an audioparam. its default value is 0, and it can range between positive and negative infinity. example in the following example, you can see an example of how the createpanner() method, audiolistener and pannernode would be used to control audio spatialisation. ...
...mylistener = audioctx.listener; mylistener.positionz.value = 1; value an audioparam. its default value is 0, and it can range between positive and negative infinity. example in the following example, you can see an example of how the createpanner() method, audiolistener and pannernode would be used to control audio spatialisation. generally you will define the position in 3d space that your audio listener and panner (source) occupy initially, and then update the position of one or both of these as the application is used. you might be moving a character around inside a game world for example, and wanting delivery of audio to change realistically as your character moves closer to or further away from a music player such as a stereo. in the example you can see this being controlled by the f...
...tion via the positionpanner() function. to see a complete implementation, check out our panner-node example (view the source code) — this demo transports you to the 2.5d "room of metal", where you can play a track on a boom box and then walk around the boom box to see how the sound changes! note how we have used some feature detection to either give the browser the newer property values (like audiolistener.forwardx) for setting position, etc. if it supports those, or older methods (like audiolistener.setorientation()) if it still supports those but not the new properties. // set up listener and panner position information var width = window.innerwidth; var height = window.innerheight; var xpos = math.floor(width/2); var ypos = math.floor(height/2); var zpos = 295; // define other variabl...
...And 11 more matches

AudioListener.setPosition() - Web APIs
the setposition() method of the audiolistener interface defines the position of the listener. the three parameters x, y and z are unitless and describe the listener's position in 3d space according to the right-hand cartesian coordinate system. pannernode objects use this position relative to individual audio sources for spatialisation. the default value of the position vector is (0, 0, 0). syntax var audioctx = new audiocontext(); var mylistener = audioctx.listener; mylistener.setposition(1,1,1); returns void. example in the following example, you can see an example of how the createpanner() method, audiolistener and pannernode would be used to control audio spatialisation. ...
...lue of the position vector is (0, 0, 0). syntax var audioctx = new audiocontext(); var mylistener = audioctx.listener; mylistener.setposition(1,1,1); returns void. example in the following example, you can see an example of how the createpanner() method, audiolistener and pannernode would be used to control audio spatialisation. generally you will define the position in 3d space that your audio listener and panner (source) occupy initially, and then update the position of one or both of these as the application is used. you might be moving a character around inside a game world for example, and wanting delivery of audio to change realistically as your character moves closer to or further away from a music player such as a stereo. in the example you can see this being controlled by the f...
...tion via the positionpanner() function. to see a complete implementation, check out our panner-node example (view the source code) — this demo transports you to the 2.5d "room of metal", where you can play a track on a boom box and then walk around the boom box to see how the sound changes! note how we have used some feature detection to either give the browser the newer property values (like audiolistener.forwardx) for setting position, etc. if it supports those, or older methods (like audiolistener.setorientation()) if it still supports those but not the new properties. // set up listener and panner position information var width = window.innerwidth; var height = window.innerheight; var xpos = math.floor(width/2); var ypos = math.floor(height/2); var zpos = 295; // define other variabl...
...And 11 more matches

AudioListener.upX - Web APIs
the upx read-only property of the audiolistener interface is an audioparam representing the x value of the direction vector defining the up direction the listener is pointing in. note: the parameter is a-rate when used with a pannernode whose pannernode is set to equalpower, or k-rate otherwise. syntax var audioctx = new audiocontext(); var mylistener = audioctx.listener; mylistener.upx.value = 0; value an audioparam. its default value is 0, and it can range between positive and negative infinity. example in the following example, you can see an example of how the createpanner() method, audiolistener and pannernode would be used to control audio spatialisation. ...
...; var mylistener = audioctx.listener; mylistener.upx.value = 0; value an audioparam. its default value is 0, and it can range between positive and negative infinity. example in the following example, you can see an example of how the createpanner() method, audiolistener and pannernode would be used to control audio spatialisation. generally you will define the position in 3d space that your audio listener and panner (source) occupy initially, and then update the position of one or both of these as the application is used. you might be moving a character around inside a game world for example, and wanting delivery of audio to change realistically as your character moves closer to or further away from a music player such as a stereo. in the example you can see this being controlled by the f...
...tion via the positionpanner() function. to see a complete implementation, check out our panner-node example (view the source code) — this demo transports you to the 2.5d "room of metal", where you can play a track on a boom box and then walk around the boom box to see how the sound changes! note how we have used some feature detection to either give the browser the newer property values (like audiolistener.forwardx) for setting position, etc. if it supports those, or older methods (like audiolistener.setorientation()) if it still supports those but not the new properties. // set up listener and panner position information var width = window.innerwidth; var height = window.innerheight; var xpos = math.floor(width/2); var ypos = math.floor(height/2); var zpos = 295; // define other variabl...
...And 11 more matches

AudioListener.upY - Web APIs
the upy read-only property of the audiolistener interface is an audioparam representing the y value of the direction vector defining the up direction the listener is pointing in. note: the parameter is a-rate when used with a pannernode whose pannernode is set to equalpower, or k-rate otherwise. syntax var audioctx = new audiocontext(); var mylistener = audioctx.listener; mylistener.upy.value = 0; value an audioparam. its default value is 1, and it can range between positive and negative infinity. example in the following example, you can see an example of how the createpanner() method, audiolistener and pannernode would be used to control audio spatialisation. ...
...; var mylistener = audioctx.listener; mylistener.upy.value = 0; value an audioparam. its default value is 1, and it can range between positive and negative infinity. example in the following example, you can see an example of how the createpanner() method, audiolistener and pannernode would be used to control audio spatialisation. generally you will define the position in 3d space that your audio listener and panner (source) occupy initially, and then update the position of one or both of these as the application is used. you might be moving a character around inside a game world for example, and wanting delivery of audio to change realistically as your character moves closer to or further away from a music player such as a stereo. in the example you can see this being controlled by the f...
...tion via the positionpanner() function. to see a complete implementation, check out our panner-node example (view the source code) — this demo transports you to the 2.5d "room of metal", where you can play a track on a boom box and then walk around the boom box to see how the sound changes! note how we have used some feature detection to either give the browser the newer property values (like audiolistener.forwardx) for setting position, etc. if it supports those, or older methods (like audiolistener.setorientation()) if it still supports those but not the new properties. // set up listener and panner position information var width = window.innerwidth; var height = window.innerheight; var xpos = math.floor(width/2); var ypos = math.floor(height/2); var zpos = 295; // define other variabl...
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AudioListener.upZ - Web APIs
the upz read-only property of the audiolistener interface is an audioparam representing the z value of the direction vector defining the up direction the listener is pointing in. note: the parameter is a-rate when used with a pannernode whose pannernode is set to equalpower, or k-rate otherwise. syntax var audioctx = new audiocontext(); var mylistener = audioctx.listener; mylistener.upz.value = 0; value an audioparam. its default value is 0, and it can range between positive and negative infinity. example in the following example, you can see an example of how the createpanner() method, audiolistener and pannernode would be used to control audio spatialisation. ...
...; var mylistener = audioctx.listener; mylistener.upz.value = 0; value an audioparam. its default value is 0, and it can range between positive and negative infinity. example in the following example, you can see an example of how the createpanner() method, audiolistener and pannernode would be used to control audio spatialisation. generally you will define the position in 3d space that your audio listener and panner (source) occupy initially, and then update the position of one or both of these as the application is used. you might be moving a character around inside a game world for example, and wanting delivery of audio to change realistically as your character moves closer to or further away from a music player such as a stereo. in the example you can see this being controlled by the f...
...tion via the positionpanner() function. to see a complete implementation, check out our panner-node example (view the source code) — this demo transports you to the 2.5d "room of metal", where you can play a track on a boom box and then walk around the boom box to see how the sound changes! note how we have used some feature detection to either give the browser the newer property values (like audiolistener.forwardx) for setting position, etc. if it supports those, or older methods (like audiolistener.setorientation()) if it still supports those but not the new properties. // set up listener and panner position information var width = window.innerwidth; var height = window.innerheight; var xpos = math.floor(width/2); var ypos = math.floor(height/2); var zpos = 295; // define other variabl...
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AudioParam.linearRampToValueAtTime() - Web APIs
the linearramptovalueattime() method of the audioparam interface schedules a gradual linear change in the value of the audioparam. the change starts at the time specified for the previous event, follows a linear ramp to the new value given in the value parameter, and reaches the new value at the time given in the endtime parameter. syntax var audioparam = audioparam.linearramptovalueattime(value, endtime) parameters value a floating point number representing the value the audioparam will ramp to by the given time. endtime a double representing the exact time (in seconds) after the ramping starts that the changing of the value will stop. returns a reference to this audioparam object. in some browsers older implementations of this interface return void. example in this example...
...esenting the value the audioparam will ramp to by the given time. endtime a double representing the exact time (in seconds) after the ramping starts that the changing of the value will stop. returns a reference to this audioparam object. in some browsers older implementations of this interface return void. example in this example, we have a media source with two control buttons (see the audio-param repo for the source code, or view the example live.) when these buttons are pressed, linearramptovalueattime() is used to fade the gain value up to 1.0, and down to 0, respectively. this is pretty useful for fade in/fade out effects, although audioparam.exponentialramptovalueattime() is often said to be a bit more natural. ...
...ve a media source with two control buttons (see the audio-param repo for the source code, or view the example live.) when these buttons are pressed, linearramptovalueattime() is used to fade the gain value up to 1.0, and down to 0, respectively. this is pretty useful for fade in/fade out effects, although audioparam.exponentialramptovalueattime() is often said to be a bit more natural. ...
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AudioParam.setValueCurveAtTime() - Web APIs
the setvaluecurveattime() method of the audioparam interface schedules the parameter's value to change following a curve defined by a list of values. the curve is a linear interpolation between the sequence of values defined in an array of floating-point values, which are scaled to fit into the given interval starting at starttime and a specific duration. syntax var paramref = param.setvaluecurveattime(values, starttime, duration); parameters values an array of floating-point numbers representing the value curve the audioparam will change through along the specified duration. every value in the array must be a finite number; if any value is nan, infinity, or -infinity, a typeerror exception is thrown. ...
...ram.setvaluecurveattime(values, starttime, duration); parameters values an array of floating-point numbers representing the value curve the audioparam will change through along the specified duration. every value in the array must be a finite number; if any value is nan, infinity, or -infinity, a typeerror exception is thrown. starttime a double representing the time (in seconds) after the audiocontext was first created that the change in value will happen. if this value is lower than audiocontext.currenttime, it is clamped to currenttime. ...
...some older browser implementations of this interface return undefined. exceptions invalidstateerror the specified array of values has fewer than 2 items in it. rangeerror the specified starttime is either negative or a non-finite value, or duration is not a finite, strictly positive number. typeerror one or more of the values in the values array is non-finite. ...
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AudioParam - Web APIs
the web audio api's audioparam interface represents an audio-related parameter, usually a parameter of an audionode (such as gainnode.gain). an audioparam can be set to a specific value or a change in value, and can be scheduled to happen at a specific time and following a specific pattern. there are two kinds of audioparam, a-rate and k-rate parameters: an a-rate audioparam takes the current audio parameter value for each sample frame of the audio signal. a k-rate audioparam uses the same initial audio parameter value for the whole block processed, that is 128 sample frames. in other words, the same value applies to every frame in the audio as it's processed by the node. each audionode defines which of its parameters are a-rate or k-rate in the spec. ...
...oparam takes the current audio parameter value for each sample frame of the audio signal. a k-rate audioparam uses the same initial audio parameter value for the whole block processed, that is 128 sample frames. in other words, the same value applies to every frame in the audio as it's processed by the node. each audionode defines which of its parameters are a-rate or k-rate in the spec. each audioparam has a list of events, initially empty, that define when and how values change. when this list is not empty, changes using the audioparam.value attributes are ignored. this list of events allows us to schedule changes that have to happen at very precise times, using arbitrary timelime-based automation curves. ...
...rs are a-rate or k-rate in the spec. each audioparam has a list of events, initially empty, that define when and how values change. when this list is not empty, changes using the audioparam.value attributes are ignored. this list of events allows us to schedule changes that have to happen at very precise times, using arbitrary timelime-based automation curves. the time used is the one defined in audiocontext.currenttime. properties audioparam.defaultvalue read only represents the initial volume of the attribute as defined by the specific audionode creating the audioparam. audioparam.maxvalue read only represents the maximum possible value for the parameter's nominal (effective) range. ...
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BaseAudioContext.createPanner() - Web APIs
the createpanner() method of the baseaudiocontext interface is used to create a new pannernode, which is used to spatialize an incoming audio stream in 3d space. the panner node is spatialized in relation to the audiocontext's audiolistener (defined by the audiocontext.listener attribute), which represents the position and orientation of the person listening to the audio. syntax baseaudioctx.createpanner(); returns a pannernode. example in the following example, you can see an example of how the createpanner() method, audiolistener and pannernode would be used to control audio spatialisation. ...
...t.listener attribute), which represents the position and orientation of the person listening to the audio. syntax baseaudioctx.createpanner(); returns a pannernode. example in the following example, you can see an example of how the createpanner() method, audiolistener and pannernode would be used to control audio spatialisation. generally you will define the position in 3d space that your audio listener and panner (source) occupy initially, and then update the position of one or both of these as the application is used. you might be moving a character around inside a game world for example, and wanting delivery of audio to change realistically as your character moves closer to or further away from a music player such as a stereo. in the example you can see this being controlled by the f...
...tion via the positionpanner() function. to see a complete implementation, check out our panner-node example (view the source code) — this demo transports you to the 2.5d "room of metal", where you can play a track on a boom box and then walk around the boom box to see how the sound changes! note how we have used some feature detection to either give the browser the newer property values (like audiolistener.forwardx) for setting position, etc. if it supports those, or older methods (like audiolistener.setorientation()) if it still supports those but not the new properties. // set up listener and panner position information var width = window.innerwidth; var height = window.innerheight; var xpos = math.floor(width/2); var ypos = math.floor(height/2); var zpos = 295; // define other variabl...
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msAudioCategory - Web APIs
non-standard this feature is non-standard and is not on a standards track. do not use it on production sites facing the web: it will not work for every user. there may also be large incompatibilities between implementations and the behavior may change in the future. the msaudiocategory property of the html <audio> element, is a read/write proprietary attribute, specific to internet explorer and microsoft edge. ...
...s non-standard and is not on a standards track. do not use it on production sites facing the web: it will not work for every user. there may also be large incompatibilities between implementations and the behavior may change in the future. the msaudiocategory property of the html <audio> element, is a read/write proprietary attribute, specific to internet explorer and microsoft edge. msaudiocategory specifies the purpose of the audio or video media, such as background audio or alerts. syntax <audio controls="controls" msaudiocategory="backgroundcapablemedia"> </audio> the msaudiocategory property offers a variety of values that can enhance the behavior of your audio-aware app. note that you must set the msaudiocategory before setting the src property in code. value include ...
... that you must set the msaudiocategory before setting the src property in code. value include a description of the property's value, including data type and what it represents. value description background capable? ...
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HTMLAudioElement - Web APIs
it's based on, and inherits properties and methods from, the htmlmediaelement interface. ...
...l="#fff" stroke="#d4dde4" stroke-width="2px" /><text x="411" y="94" font-size="12px" font-family="consolas,monaco,andale mono,monospace" fill="#4d4e53" text-anchor="middle" alignment-baseline="middle">htmlmediaelement</text></a><polyline points="331,89 321,84 321,94 331,89" stroke="#d4dde4" fill="none"/><line x1="321" y1="89" x2="291" y2="89" stroke="#d4dde4"/><a xlink:href="/docs/web/api/htmlaudioelement" target="_top"><rect x="131" y="65" width="160" height="50" fill="#f4f7f8" stroke="#d4dde4" stroke-width="2px" /><text x="211" y="94" font-size="12px" font-family="consolas,monaco,andale mono,monospace" fill="#4d4e53" text-anchor="middle" alignment-baseline="middle">htmlaudioelement</text></a></svg></div> a:hover text { fill: #0095dd; pointer-events: all;} constructor audio()...
...131" y="65" width="160" height="50" fill="#f4f7f8" stroke="#d4dde4" stroke-width="2px" /><text x="211" y="94" font-size="12px" font-family="consolas,monaco,andale mono,monospace" fill="#4d4e53" text-anchor="middle" alignment-baseline="middle">htmlaudioelement</text></a></svg></div> a:hover text { fill: #0095dd; pointer-events: all;} constructor audio() creates and returns a new htmlaudioelement object, optionally starting the process of loading an audio file into it if the file url is given. properties no specific properties; inherits properties from its parent, htmlmediaelement, and from htmlelement. methods inherits methods from its parent, htmlmediaelement, and from htmlelement. it offers no methods of its own. obsolete mozilla-only methods the following methods are no...
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AudioContext.createMediaStreamTrackSource() - Web APIs
the createmediastreamtracksource() method of the audiocontext interface creates and returns a mediastreamtrackaudiosourcenode which represents an audio source whose data comes from the specified mediastreamtrack. this differs from createmediastreamsource(), which creates a mediastreamaudiosourcenode whose audio comes from the audio track in a specified mediastream whose id is first, lexicographically (alphabetically). syntax var audioctx = new audiocontext(); var track = audioctx.createmediastreamtracksource(track); parameters track the mediastreamtrack to use as the source of all audio data for the new node. return value a mediastreamtrackaudiosourcenode object which acts as a source for audio data found in the specified audio track. example in this example, getusermedia() is...
...octx.createmediastreamtracksource(track); parameters track the mediastreamtrack to use as the source of all audio data for the new node. return value a mediastreamtrackaudiosourcenode object which acts as a source for audio data found in the specified audio track. example in this example, getusermedia() is used to request access to the user's microphone. once that access is attained, an audio context is established and a mediastreamtrackaudiosourcenode is created using createmediastreamtracksource(), taking its audio from the first audio track in the stream returned by getusermedia(). then a biquadfilternode is created using createbiquadfilter(), and it's configured as desired to perform a lowshelf filter on the audio coming from the source. the output from the microphone is then rou...
...eatemediastreamtracksource(), taking its audio from the first audio track in the stream returned by getusermedia(). then a biquadfilternode is created using createbiquadfilter(), and it's configured as desired to perform a lowshelf filter on the audio coming from the source. the output from the microphone is then routed into the new biquad filter, and the filter's output is in turn routed to the audio context's destination. ...
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AudioListener.dopplerFactor - Web APIs
the deprecated dopplerfactor property of the audiolistener interface is a double value representing the amount of pitch shift to use when rendering a doppler effect. the dopplerfactor property's default value is 1, which is a sensible default for most situations. syntax var audioctx = new audiocontext(); var mylistener = audioctx.listener; mylistener.dopplerfactor = 1; value a double indicating the doppler effect's pitch shift value. the value is 1 by default. example in the following example, you can see an example of how the createpanner() method, audiolistener and pannernode would be used to control audio spatialisation. ...
...ext(); var mylistener = audioctx.listener; mylistener.dopplerfactor = 1; value a double indicating the doppler effect's pitch shift value. the value is 1 by default. example in the following example, you can see an example of how the createpanner() method, audiolistener and pannernode would be used to control audio spatialisation. generally you will define the position in 3d space that your audio listener and panner (source) occupy initially, and then update the position of one or both of these as the application is used. you might be moving a character around inside a game world for example, and wanting delivery of audio to change realistically as your character moves closer to or further away from a music player such as a stereo. in the example you can see this being controlled by the f...
...tion via the positionpanner() function. to see a complete implementation, check out our panner-node example (view the source code) — this demo transports you to the 2.5d "room of metal", where you can play a track on a boom box and then walk around the boom box to see how the sound changes! note how we have used some feature detection to either give the browser the newer property values (like audiolistener.forwardx) for setting position, etc. if it supports those, or older methods (like audiolistener.setorientation()) if it still supports those but not the new properties. // set up listener and panner position information var width = window.innerwidth; var height = window.innerheight; var xpos = math.floor(width/2); var ypos = math.floor(height/2); var zpos = 295; // define other variabl...
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AudioListener.speedOfSound - Web APIs
the speedofsound property of the audiolistener interface is a double value representing the speed of sound, in meters per second. the speedofsound property's default value is 343.3 m/s and is used to calculate the doppler shift appropriate for the speed the panner is travelling at (as defined by pannernode.setvelocity.) note: bear in mind that no propagation delay is automatically applied to a sound far from the listener. syntax var audioctx = new audiocontext(); var mylistener = audioctx.listener; mylistener.speedofsound = 343.3; value a double. example in the following example, you can see an example of how the createpanner() method, audiolistener and pannernode would be used to control audio spatialisation. ...
...d to a sound far from the listener. syntax var audioctx = new audiocontext(); var mylistener = audioctx.listener; mylistener.speedofsound = 343.3; value a double. example in the following example, you can see an example of how the createpanner() method, audiolistener and pannernode would be used to control audio spatialisation. generally you will define the position in 3d space that your audio listener and panner (source) occupy initially, and then update the position of one or both of these as the application is used. you might be moving a character around inside a game world for example, and wanting delivery of audio to change realistically as your character moves closer to or further away from a music player such as a stereo. in the example you can see this being controlled by the f...
...tion via the positionpanner() function. to see a complete implementation, check out our panner-node example (view the source code) — this demo transports you to the 2.5d "room of metal", where you can play a track on a boom box and then walk around the boom box to see how the sound changes! note how we have used some feature detection to either give the browser the newer property values (like audiolistener.forwardx) for setting position, etc. if it supports those, or older methods (like audiolistener.setorientation()) if it still supports those but not the new properties. // set up listener and panner position information var width = window.innerwidth; var height = window.innerheight; var xpos = math.floor(width/2); var ypos = math.floor(height/2); var zpos = 295; // define other variabl...
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AudioParam.exponentialRampToValueAtTime() - Web APIs
the exponentialramptovalueattime() method of the audioparam interface schedules a gradual exponential change in the value of the audioparam. the change starts at the time specified for the previous event, follows an exponential ramp to the new value given in the value parameter, and reaches the new value at the time given in the endtime parameter. note: exponential ramps are considered more useful when changing frequencies or playback rates than linear ramps because of the way the human ear works. syntax var audioparam = audioparam.exponentialramptovalueattime(value, endtime) parameters value a floating point number representing the value the audioparam will ramp to by the given time. endtime a double representing the exact time (in seconds) after the ramping starts that the chan...
...ps because of the way the human ear works. syntax var audioparam = audioparam.exponentialramptovalueattime(value, endtime) parameters value a floating point number representing the value the audioparam will ramp to by the given time. endtime a double representing the exact time (in seconds) after the ramping starts that the changing of the value will stop. returns a reference to this audioparam object. in some browsers older implementations of this interface return void. examples in this example, we have a media source with two control buttons (see the audio-param repo for the source code, or view the example live.) when these buttons are pressed, exponentialramptovalueattime() is used to fade the gain value up to 1.0, and down to 0, respectively. this is pretty useful for fade i...
...older implementations of this interface return void. examples in this example, we have a media source with two control buttons (see the audio-param repo for the source code, or view the example live.) when these buttons are pressed, exponentialramptovalueattime() is used to fade the gain value up to 1.0, and down to 0, respectively. ...
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AudioWorkletProcessor.process - Web APIs
the process() method of an audioworkletprocessor-derived class implements the audio processing algorithm for the audio processor worklet. although the method is not a part of the audioworkletprocessor interface, any implementation of audioworkletprocessor must provide a process() method. the method is called synchronously from the audio rendering thread, once for each block of audio (also known as a rendering quantum) being directed through the processor's corresponding audioworkletnode. in other words, every time a new block of audio is ready for your processor to manipulate, your process() function is invoked to do so. important: currently, audio data blocks are always 128 frames long—that is, they contain 128 32-bit floating-point samples for each of the inputs' channels. however, plan...
...therefore, you must always check the size of the sample array rather than assuming a particular size. ...
...your processor to manipulate, your process() function is invoked to do so. important: currently, audio data blocks are always 128 frames long—that is, they contain 128 32-bit floating-point samples for each of the inputs' channels. however, plans are already in place to revise the specification to allow the size of the audio blocks to be changed depending on circumstances (for example, if the audio hardware or cpu utilization is more efficient with larger block sizes). therefore, you must always check the size of the sample array rather than assuming a particular size. ...
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AudioWorkletProcessor - Web APIs
the audioworkletprocessor interface of the web audio api represents an audio processing code behind a custom audioworkletnode. it lives in the audioworkletglobalscope and runs on the web audio rendering thread. in turn, an audioworkletnode based on it runs on the main thread. constructor the audioworkletprocessor and classes that derive from it cannot be instantiated directly from a user-supplied code. instead, they are created only internally by the creation of an associated audioworkletnodes. the constructor of the deriving class is getting called with an options object, so you can perform a custom initialization procedures — see constructor page for details. audioworkletprocessor() creates a new instance of an audioworkletprocessor object. properties port read only returns a ...
...audioworkletnodes. the constructor of the deriving class is getting called with an options object, so you can perform a custom initialization procedures — see constructor page for details. audioworkletprocessor() creates a new instance of an audioworkletprocessor object. properties port read only returns a messageport used for bidirectional communication between the processor and the audioworkletnode which it belongs to. the other end is available under the port property of the node. methods the audioworkletprocessor interface does not define any methods of its own. however, you must provide a process() method, which is called in order to process the audio stream. events the audioworkletprocessor interface doesn't respond to any events. usage notes deriving classes to defin...
...de which it belongs to. the other end is available under the port property of the node. methods the audioworkletprocessor interface does not define any methods of its own. however, you must provide a process() method, which is called in order to process the audio stream. events the audioworkletprocessor interface doesn't respond to any events. usage notes deriving classes to define custom audio processing code you have to derive a class from the audioworkletprocessor interface. although not defined on the interface, the deriving class must have the process method. ...
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Audio() - Web APIs
if a url is specified, the browser begins to asynchronously load the media resource before returning the new object. ...
...if a url is specified, the browser begins to asynchronously load the media resource before returning the new object. ...
...if a url is specified, the browser begins to asynchronously load the media resource before returning the new object. ...
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ScriptProcessorNode.onaudioprocess - Web APIs
note: as of the august 29 2014 web audio api spec publication, this feature has been marked as deprecated, and is soon to be replaced by audio workers. the onaudioprocess event handler of the scriptprocessornode interface represents the eventhandler to be called for the audioprocess event that is dispatched to scriptprocessornode node types. an event of type audioprocessingevent will be dispatched to the event handler. syntax var audioctx = new audiocontext(); var scriptnode = audioctx.createscriptprocessor(4096, 1, 1); scriptnode.onaudioprocess = function() { ... ...
...ocess event that is dispatched to scriptprocessornode node types. an event of type audioprocessingevent will be dispatched to the event handler. syntax var audioctx = new audiocontext(); var scriptnode = audioctx.createscriptprocessor(4096, 1, 1); scriptnode.onaudioprocess = function() { ... } example the following example shows basic usage of a scriptprocessornode to take a track loaded via audiocontext.decodeaudiodata(), process it, adding a bit of white noise to each audio sample of the input track (buffer) and play it through the audiodestinationnode. ...
...iptprocessor(4096, 1, 1); scriptnode.onaudioprocess = function() { ... } example the following example shows basic usage of a scriptprocessornode to take a track loaded via audiocontext.decodeaudiodata(), process it, adding a bit of white noise to each audio sample of the input track (buffer) and play it through the audiodestinationnode. for each channel and each sample frame, the scriptnode.onaudioprocess function takes the associated audioprocessingevent and uses it to loop through each channel of the input buffer, and each sample in each channel, and add a small amount of white noise, before setting that result to be the output sample in each case. note: for a full working example, see our script-processor-node github repo (also view the source code.) var myscript = document.querysele...
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Web audio spatialization basics - Web APIs
think sounds flying over you, creeping up behind you, moving across in front of you. ...
...ing (and other) options wasn't enough, the web audio api also includes facilities to allow you to emulate the difference in sound as a listener moves around a sound source, for example panning as you move around a sound source inside a 3d game. the official term for this is spatialization, and this article will cover the basics of how to implement such a system. basics of spatialization in web audio, complex 3d spatializations are created using the pannernode, which in layman's terms is basically a whole lotta cool maths to make audio appear in 3d space. think sounds flying over you, creeping up behind you, moving across in front of you. that sort of thing. it's really useful for webxr and gaming. in 3d spaces, it's the only way to achieve realistic audio. libraries like three.js and a-fram...
...reating simple left and right stereo panning effects. this is much simpler to use, but obviously nowhere near as versatile. if you just want a simple stereo panning effect, our stereopannernode example (see source code) should give you everything you need. 3d boombox demo to demonstrate 3d spatialization we've created a modified version of the boombox demo we created in our basic using the web audio api guide. see the 3d spatialization demo live (and see the source code also). the boombox sits inside a room (defined by the edges of the browser viewport), and in this demo, we can move and rotate it with the provided controls. when we move the boombox, the sound it produces changes accordingly, panning as it moves to the left or right of the room, or becoming quieter as it is moved away fro...
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AudioWorkletGlobalScope.registerProcessor - Web APIs
a new processor by the given name is internally created and associated with the new node. ...
...a new processor by the given name is internally created and associated with the new node. ...
...a new processor by the given name is internally created and associated with the new node. ...
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BaseAudioContext.createBiquadFilter() - Web APIs
the createbiquadfilter() method of the baseaudiocontext interface creates a biquadfilternode, which represents a second order filter configurable as several different common filter types. syntax baseaudiocontext.createbiquadfilter(); returns a biquadfilternode. example the following example shows basic usage of an audiocontext to create a biquad filter node. for a complete working example, check out our voice-change-o-matic demo (look at the source code too). ...
...biquadfilternode, which represents a second order filter configurable as several different common filter types. syntax baseaudiocontext.createbiquadfilter(); returns a biquadfilternode. example the following example shows basic usage of an audiocontext to create a biquad filter node. for a complete working example, check out our voice-change-o-matic demo (look at the source code too). ...
... represents a second order filter configurable as several different common filter types. syntax baseaudiocontext.createbiquadfilter(); returns a biquadfilternode. example the following example shows basic usage of an audiocontext to create a biquad filter node. for a complete working example, check out our voice-change-o-matic demo (look at the source code too). ...
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ScriptProcessorNode: audioprocess event - Web APIs
the audioprocess event of the scriptprocessornode interface is fired when an input buffer of a script processor is ready to be processed. ...
... the audioprocess event of the scriptprocessornode interface is fired when an input buffer of a script processor is ready to be processed. ...
... the audioprocess event of the scriptprocessornode interface is fired when an input buffer of a script processor is ready to be processed. ...
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Visualizations with Web Audio API - Web APIs
one of the most interesting features of the web audio api is the ability to extract frequency, waveform, and other data from your audio source, which can then be used to create visualizations. this article explains how, and provides a couple of basic use cases. note: you can find working examples of all the code snippets in our voice-change-o-matic demo. ...
...of the most interesting features of the web audio api is the ability to extract frequency, waveform, and other data from your audio source, which can then be used to create visualizations. this article explains how, and provides a couple of basic use cases. note: you can find working examples of all the code snippets in our voice-change-o-matic demo. ...
...y to extract frequency, waveform, and other data from your audio source, which can then be used to create visualizations. this article explains how, and provides a couple of basic use cases. note: you can find working examples of all the code snippets in our voice-change-o-matic demo. ...
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AudioContext - Web APIs
the audiocontext interface represents an audio-processing graph built from audio modules linked together, each represented by an audionode. an audio context controls both the creation of the nodes it contains and the execution of the audio processing, or decoding. you need to create an audiocontext before you do anything else, as everything happens inside a context. it's recommended to create one audiocontext and reuse it instead of initializing a new one each time, and it's ok to use a single audiocontext for several different audio source and pipeline concurrently. <div id="interfacediagram" style="display: inline-block; position: relative; width: 100%; padding-bottom: 11.666666666666666%; vertical-align: middle; overflow: hidden;"><svg style="display: inline-block; position: absolute; ...
...t="50" fill="#fff" stroke="#d4dde4" stroke-width="2px" /><text x="56" y="30" font-size="12px" font-family="consolas,monaco,andale mono,monospace" fill="#4d4e53" text-anchor="middle" alignment-baseline="middle">eventtarget</text></a><polyline points="111,25 121,20 121,30 111,25" stroke="#d4dde4" fill="none"/><line x1="121" y1="25" x2="151" y2="25" stroke="#d4dde4"/><a xlink:href="/docs/web/api/audiocontext" target="_top"><rect x="151" y="1" width="120" height="50" fill="#f4f7f8" stroke="#d4dde4" stroke-width="2px" /><text x="211" y="30" font-size="12px" font-family="consolas,monaco,andale mono,monospace" fill="#4d4e53" text-anchor="middle" alignment-baseline="middle">audiocontext</text></a></svg></div> a:hover text { fill: #0095dd; pointer-events: all;} constructor audiocontext...
...="151" y="1" width="120" height="50" fill="#f4f7f8" stroke="#d4dde4" stroke-width="2px" /><text x="211" y="30" font-size="12px" font-family="consolas,monaco,andale mono,monospace" fill="#4d4e53" text-anchor="middle" alignment-baseline="middle">audiocontext</text></a></svg></div> a:hover text { fill: #0095dd; pointer-events: all;} constructor audiocontext() creates and returns a new audiocontext object. properties also inherits properties from its parent interface, baseaudiocontext. ...
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AudioTrack - Web APIs
the id can also be used as the fragment part of the url if the media supports seeking by media fragment per the media fragments uri specification. ...
...the id can also be used as the fragment part of the url if the media supports seeking by media fragment per the media fragments uri specification. ...
...the id can also be used as the fragment part of the url if the media supports seeking by media fragment per the media fragments uri specification. ...
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BaseAudioContext.createDynamicsCompressor() - Web APIs
the createdynamicscompressor() method of the baseaudiocontext interface is used to create a dynamicscompressornode, which can be used to apply compression to an audio signal. compression lowers the volume of the loudest parts of the signal and raises the volume of the softest parts. overall, a louder, richer, and fuller sound can be achieved. it is especially important in games and musical applications where large numbers of individual sounds are played simultaneously, where you want to control the overall signal level and help avoid clipping (distorting) of the audio output. syntax baseaudioctx.createdynamicscompressor(); returns a dynamicscompressornode. example the below code demonstrates a simple usage of createdynamicscompressor() to add compression to an audio track. for a more ...
...signal level and help avoid clipping (distorting) of the audio output. syntax baseaudioctx.createdynamicscompressor(); returns a dynamicscompressornode. example the below code demonstrates a simple usage of createdynamicscompressor() to add compression to an audio track. for a more complete example, have a look at our basic compressor example (view the source code). ...
...put. syntax baseaudioctx.createdynamicscompressor(); returns a dynamicscompressornode. example the below code demonstrates a simple usage of createdynamicscompressor() to add compression to an audio track. for a more complete example, have a look at our basic compressor example (view the source code). ...
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BaseAudioContext.createStereoPanner() - Web APIs
the createstereopanner() method of the baseaudiocontext interface creates a stereopannernode, which can be used to apply stereo panning to an audio source. it positions an incoming audio stream in a stereo image using a low-cost equal-power panning algorithm. syntax baseaudiocontext.createstereopanner(); returns a stereopannernode. example in our stereopannernode example (see source code) html we have a simple <audio> element along with a slider <input> to increase and decrease pan value. in the javascript we create a mediaelementaudiosourcenode and a stereopannernode, and connect the two together using the connect() method. we then use an oninput event handler to change the value of the stereopannernode.pan parameter and update the pan value display when the slider is moved. moving t...
...taudiosourcenode and a stereopannernode, and connect the two together using the connect() method. we then use an oninput event handler to change the value of the stereopannernode.pan parameter and update the pan value display when the slider is moved. moving the slider left and right while the music is playing pans the music across to the left and right speakers of the output, respectively. ...
...stereopannernode, and connect the two together using the connect() method. we then use an oninput event handler to change the value of the stereopannernode.pan parameter and update the pan value display when the slider is moved. moving the slider left and right while the music is playing pans the music across to the left and right speakers of the output, respectively. ...
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nsIDOMHTMLAudioElement
do not use it on production sites facing the web: it will not work for every user. ...
...do not use it on production sites facing the web: it will not work for every user. ...
...dio> element. dom/interfaces/html/nsidomhtmlaudioelement.idlscriptable please add a summary to this article. last changed in gecko 2.0 (firefox 4 / thunderbird 3.3 / seamonkey 2.1) inherits from: nsidomhtmlmediaelement method overview unsigned long long mozcurrentsampleoffset(); void mozsetup(in pruint32 channels, in pruint32 rate); [implicit_jscontext] unsigned long mozwriteaudio(in jsval data); methods mozcurrentsampleoffset() non-standard this feature is non-standard and is not on a standards track. do not use it on production sites facing the web: it will not work for every user. there may also be large incompatibilities between implementations and the behavior may change in the future. ...
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AudioBuffer.getChannelData() - Web APIs
the getchanneldata() method of the audiobuffer interface returns a float32array containing the pcm data associated with the channel, defined by the channel parameter (with 0 representing the first channel). syntax var myarraybuffer = audioctx.createbuffer(2, framecount, audioctx.samplerate); var nowbuffering = myarraybuffer.getchanneldata(channel); parameters channel the channel property is an index representing the particular channel to get data for. an index value of 0 represents the first channel. if the channel index value is greater than of equal to audiobuffer.numberofchannels, an index_size_err exception will be thrown. return value a float32array. ...
...erty is an index representing the particular channel to get data for. an index value of 0 represents the first channel. if the channel index value is greater than of equal to audiobuffer.numberofchannels, an index_size_err exception will be thrown. return value a float32array. example in the following example we create a two second buffer, fill it with white noise, and then play it via an audiobuffersourcenode. the comments should clearly explain what is going on. you can also run the code live, or view the source. ...
... index value is greater than of equal to audiobuffer.numberofchannels, an index_size_err exception will be thrown. return value a float32array. example in the following example we create a two second buffer, fill it with white noise, and then play it via an audiobuffersourcenode. the comments should clearly explain what is going on. you can also run the code live, or view the source. ...
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AudioParam.setTargetAtTime() - Web APIs
timeconstant the time-constant value, given in seconds, of an exponential approach to the target value. ...
...rt of a gradual change to the audioparam value. this is useful for decay or release portions of adsr envelopes. syntax var paramref = param.settargetattime(target, starttime, timeconstant); parameters target the value the parameter will start to transition towards at the given start time. starttime the time that the exponential transition will begin, in the same time coordinate system as audiocontext.currenttime. if it is less than or equal to audiocontext.currenttime, the parameter will start changing immediately. timeconstant the time-constant value, given in seconds, of an exponential approach to the target value. the larger this value is, the slower the transition will be. returns a reference to this audioparam object. some older browser implementations of this interface retur...
...ven by the target parameter. the decay rate as defined by the timeconstant parameter is exponential; therefore the value will never reach target completely, but after each timestep of length timeconstant, the value will have approached target by another 1-e-1≈63.2%1 - e^{-1} \approx 63.2%. for the complete formula (which uses a first-order linear continuous time-invariant system), check the web audio specification. if you absolutely need to reach the target value by a specific time, you can use audioparam.exponentialramptovalueattime(). however, for mathematical reasons, that method does not work if the current value or the target value is 0. choosing a good timeconstant as mentioned above, the value changes exponentially, with each timeconstant bringing you another 63.2% toward the target...
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AudioTrack.enabled - Web APIs
the audiotrack property enabled specifies whether or not the described audio track is currently enabled for use. if the track is disabled by setting enabled to false, the track is muted and does not produce audio. syntax isaudioenabled = audiotrack.enabled; audiotrack.enabled = true | false; value the enabled property is a boolean whose value is true if the track is enabled; enabled tracks produce audio while the media is playing. setting enabled to false effectively mutes the audio track, preventing it from contributing to the media's audio performance. example this example switches between the main and commentary audio tracks of a media element. ...
...f the track is enabled; enabled tracks produce audio while the media is playing. setting enabled to false effectively mutes the audio track, preventing it from contributing to the media's audio performance. example this example switches between the main and commentary audio tracks of a media element. ...
...; enabled tracks produce audio while the media is playing. setting enabled to false effectively mutes the audio track, preventing it from contributing to the media's audio performance. example this example switches between the main and commentary audio tracks of a media element. ...
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AudioTrackList - Web APIs
onchange an event handler to be called when the change event occurs. ...
... onchange an event handler to be called when the change event occurs. ...
... onchange an event handler to be called when the change event occurs. ...
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AudioWorkletNode.port - Web APIs
the read-only port property of the audioworkletnode interface returns the associated messageport. it can be used to communicate between the node and its associated audioworkletprocessor. note: the port at the other end of the channel is available under the port property of the processor. syntax audioworkletnodeinstance.port; value the messageport object that is connecting the audioworkletnode and its associated audioworkletprocessor. examples to demonstrate bidirectional communication capabilities, we'll create an audioworkletprocessor, which will output silence and respond to ping requests from its audioworkletnode. first, we need to define a custom audioworkletprocessor, and register it. note that that this should be done in a separate file. // ping-pong-processor.js cla...
...udioworkletprocessor. examples to demonstrate bidirectional communication capabilities, we'll create an audioworkletprocessor, which will output silence and respond to ping requests from its audioworkletnode. first, we need to define a custom audioworkletprocessor, and register it. note that that this should be done in a separate file. ...
...ngprocessor extends audioworkletprocessor { constructor (...args) { super(...args) this.port.onmessage = (e) => { console.log(e.data) this.port.postmessage('pong') } } process (inputs, outputs, parameters) { return true } } registerprocessor('ping-pong-processor', pingpongprocessor) now in our main scripts file we'll load the processor, create an instance of audioworkletnode passing the name of the processor, and connect the node to an audio graph. ...
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AudioWorkletProcessor.port - Web APIs
the read-only port property of the audioworkletprocessor interface returns the associated messageport. it can be used to communicate between the processor and the audioworkletnode to which it belongs. note: the port at the other end of the channel is available under the port property of the node. syntax audioworkletprocessorinstance.port; value the messageport object that is connecting the audioworkletprocessor and the associated audioworkletnode. examples to demonstrate bidirectional communication capabilities, we'll create an audioworkletprocessor, which will output silence and respond to ping requests from its audioworkletnode. first, we need to define a custom audioworkletprocessor, and register it. note that that this should be done in a separate file. // ping-pong-pr...
...ted audioworkletnode. examples to demonstrate bidirectional communication capabilities, we'll create an audioworkletprocessor, which will output silence and respond to ping requests from its audioworkletnode. first, we need to define a custom audioworkletprocessor, and register it. note that that this should be done in a separate file. ...
...ngprocessor extends audioworkletprocessor { constructor (...args) { super(...args) this.port.onmessage = (e) => { console.log(e.data) this.port.postmessage('pong') } } process (inputs, outputs, parameters) { return true } } registerprocessor('ping-pong-processor', pingpongprocessor) now in our main scripts file we'll load the processor, create an instance of audioworkletnode passing the name of the processor, and connect the node to an audio graph. ...
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BaseAudioContext.createDelay() - Web APIs
must be less than 180 seconds, and defaults to 1 second if not specified. returns a delaynode. the default delaynode.delaytime is 0 seconds. example we have created a simple example that allows you to play three different samples on a constant loop — see create-delay (you can also view the source code). ...
...must be less than 180 seconds, and defaults to 1 second if not specified. returns a delaynode. the default delaynode.delaytime is 0 seconds. example we have created a simple example that allows you to play three different samples on a constant loop — see create-delay (you can also view the source code). ...
... have created a simple example that allows you to play three different samples on a constant loop — see create-delay (you can also view the source code). if you just press the play buttons, the loops will start immediately; if you slide the sliders up to the right, then press the play buttons, a delay will be introduced, so the looping sounds don't start playing for a short amount of time. ...
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MediaStreamTrackAudioSourceNode() - Web APIs
the web audio api's mediastreamtrackaudiosourcenode() constructor creates and returns a new mediastreamtrackaudiosourcenode object whose audio is taken from the mediastreamtrack specified in the given options object. another way to create a mediastreamtrackaudiosourcenode is to call theaudiocontext.createmediastreamtracksource() method, specifying the mediastreamtrack from which you want to obtain audio. syntax audiotracknode = new mediastreamtrackaudiosourcenode(context, options); parameters context an audiocontext representing the audio context you want the node to be associated with. ...
...create a mediastreamtrackaudiosourcenode is to call theaudiocontext.createmediastreamtracksource() method, specifying the mediastreamtrack from which you want to obtain audio. syntax audiotracknode = new mediastreamtrackaudiosourcenode(context, options); parameters context an audiocontext representing the audio context you want the node to be associated with. options a mediastreamtrackaudiosourceoptions object defining the properties you want the mediastreamtrackaudiosourcenode to have: mediastreamtrack the mediastreamtrack from which to take audio data for this node's output. return value a new mediastreamtrackaudiosourcenode object representing the audio node whose media is obtained from the specified media track. exceptions notsupportederror the specified context ...
...ons object defining the properties you want the mediastreamtrackaudiosourcenode to have: mediastreamtrack the mediastreamtrack from which to take audio data for this node's output. return value a new mediastreamtrackaudiosourcenode object representing the audio node whose media is obtained from the specified media track. exceptions notsupportederror the specified context is not an audiocontext. invalidstateerror the specified mediastreamtrack isn't an audio track (that is, its kind property isn't audio. example this example uses getusermedia() to obtain access to the user's camera, then creates a new mediastreamaudiosourcenode from the first audio track provided by the device. ...
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MediaStreamTrackAudioSourceNode - Web APIs
the mediastreamtrackaudiosourcenode interface is a type of audionode which represents a source of audio data taken from a specific mediastreamtrack obtained through the webrtc or media capture and streams apis. the audio itself might be input from a microphone or other audio sampling device, or might be received through a rtcpeerconnection, among other posible options. a mediastreamtrackaudiosourcenode has no inputs and exactly one output, and is created using the audiocontext.createmediastreamtracksource() method. this interface is similar to mediastreamaudiosourcenode, except it lets you specifically state the track to use, rather than assuming the first audio track on a stream. number of inputs 0 number of outputs 1 channel count defined by the first ...
...node has no inputs and exactly one output, and is created using the audiocontext.createmediastreamtracksource() method. this interface is similar to mediastreamaudiosourcenode, except it lets you specifically state the track to use, rather than assuming the first audio track on a stream. number of inputs 0 number of outputs 1 channel count defined by the first audio mediastreamtrack passed to the audiocontext.createmediastreamtracksource() method that created it. constructor new mediastreamtrackaudiosourcenode() creates a new mediastreamtrackaudiosourcenode object instance with the specified options. ...
... number of inputs 0 number of outputs 1 channel count defined by the first audio mediastreamtrack passed to the audiocontext.createmediastreamtracksource() method that created it. constructor new mediastreamtrackaudiosourcenode() creates a new mediastreamtrackaudiosourcenode object instance with the specified options. properties the mediastreamtrackaudiosourcenode interface has no properties of its own; however, it inherits the properties of its parent, audionode. methods inherits methods from its parent, audionode. 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, ...
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Writing Web Audio API code that works in every browser - Developer guides
you probably have already read the announcement on the web audio api coming to firefox, and are totally excited and ready to make your until-now-webkit-only sites work with firefox, which uses the unprefixed version of the spec. unfortunately, chrome, safari and opera still use the webkitaudiocontext prefixed name. furthermore, as a result of the spec being still in flux, some browsers use deprecated properties and method names that are not present in standards-compliant browsers: safari uses the old method names, firefox uses the new ones, and chrome and opera use both. in addition, not all features of web audio are already implemented in firefox yet. what do we do!? we don't want to maintain two or more separate code bases, and feature detection code is cumbersome! plus we want to wri...
...e already implemented in firefox yet. what do we do!? we don't want to maintain two or more separate code bases, and feature detection code is cumbersome! plus we want to write code that reliably works in the future, or at least, works with a minimum amount of changes. ...
...also, if new methods such as start are not detected in some nodes, the library will also alias them to their old names. thus, start is mapped to noteon, stop to noteoff, and so on. ...
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AudioBufferSourceNode.loop - Web APIs
the loop property of the audiobuffersourcenode interface is a boolean indicating if the audio asset must be replayed when the end of the audiobuffer is reached. the loop property's default value is false. syntax var loopingenabled = audiobuffersourcenode.loop; audiobuffersourcenode.loop = true | false; value a boolean which is true if looping is enabled; otherwise, the value is false. when looping is enabled, the sound begins playing at the time specified as the start point when start() is called. when the time specified by the loopend property is reached, playback continues at the time specified by loopstart example in this example, the audiocontext.decodeaudiodata function is used to decode an audio track and put it into an audiobuffersourcenode. buttons are provided to play and s...
... enabled, the sound begins playing at the time specified as the start point when start() is called. when the time specified by the loopend property is reached, playback continues at the time specified by loopstart example in this example, the audiocontext.decodeaudiodata function is used to decode an audio track and put it into an audiobuffersourcenode. buttons are provided to play and stop the audio playback, and a slider control is used to change the playbackrate property value on the fly. when the audio is played, it loops. ...
...this example, the audiocontext.decodeaudiodata function is used to decode an audio track and put it into an audiobuffersourcenode. buttons are provided to play and stop the audio playback, and a slider control is used to change the playbackrate property value on the fly. when the audio is played, it loops. ...
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AudioBufferSourceNode.loopEnd - Web APIs
the loopend property of the audiobuffersourcenode interface specifies is a floating point number specifying, in seconds, at what offset into playing the audiobuffer playback should loop back to the time indicated by the loopstart property. this is only used if the loop property is true. syntax audiobuffersourcenode.loopend = endoffsetinseconds; var endoffsetinseconds = audiobuffersourcenode.loopend; value a floating-point number indicating the offset, in seconds, into the audio buffer at which each loop will loop return to the beginning of the loop (that is, the current play time gets reset to audiobuffersourcenode.loopstart). this property is only used if the loop property is true. the default value is 0. ...
...endoffsetinseconds = audiobuffersourcenode.loopend; value a floating-point number indicating the offset, in seconds, into the audio buffer at which each loop will loop return to the beginning of the loop (that is, the current play time gets reset to audiobuffersourcenode.loopstart). this property is only used if the loop property is true. the default value is 0. example in this example, the audiocontext.decodeaudiodata() function is used to decode an audio track and put it into an audiobuffersourcenode. buttons are provided to play and stop the audio playback, and slider controls are used to change the playbackrate, loopstart and loopend properties on the fly. when the audio is played to the end, it loops, but you can control how long the loops last by altering loopstart and loopend. for...
...values to 20 and 25, respectively, then begin playback, the sound will play normally until it reaches the 25 second mark. then the current play position will loop back to the 20 second mark and continue playing until the 25 second mark, ad infinitum (or at least until stop() is called). for a full working example, see this code running live, or view the source. ...
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AudioBufferSourceNode.loopStart - Web APIs
the loopstart property of the audiobuffersourcenode interface is a floating-point value indicating, in seconds, where in the audiobuffer the restart of the play must happen. the loopstart property's default value is 0. syntax audiobuffersourcenode.loopstart = startoffsetinseconds; startoffsetinseconds = audiobuffersourcenode.loopstart; value a floating-point number indicating the offset, in seconds, into the audio buffer at which each loop should begin during playback. this value is only used when the loop parameter is true. example in this example, the audiocontext.decodeaudiodata() function is used to decode an audio track and put it into an audiobuffersourcenode. ...
...node.loopstart; value a floating-point number indicating the offset, in seconds, into the audio buffer at which each loop should begin during playback. this value is only used when the loop parameter is true. example in this example, the audiocontext.decodeaudiodata() function is used to decode an audio track and put it into an audiobuffersourcenode. buttons are provided to play and stop the audio playback, and slider controls are used to change the playbackrate, loopstart, and loopend properties on the fly. when the audio is played to the end, it loops, but you can control how long the loops last by altering loopstart and loopend. for example, if you set their values to 20 and 25, respectively, the audio will start to loop between 20 and 25 seconds in to the track. for a full working e...
... properties on the fly. when the audio is played to the end, it loops, but you can control how long the loops last by altering loopstart and loopend. for example, if you set their values to 20 and 25, respectively, the audio will start to loop between 20 and 25 seconds in to the track. for a full working example, see this code running live, or view the source. ...
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AudioContext.createMediaStreamSource() - Web APIs
the createmediastreamsource() method of the audiocontext interface is used to create a new mediastreamaudiosourcenode object, given a media stream (say, from a mediadevices.getusermedia instance), the audio from which can then be played and manipulated. for more details about media stream audio source nodes, check out the mediastreamaudiosourcenode reference page. syntax audiosourcenode = audiocontext.createmediastreamsource(stream); parameters stream a mediastream to serve as an audio source to be fed into an audio processing graph for use and manipulation. return value a new mediastreamaudiosourcenode object representing the audio node whose media is obtained from the specified source stream. ...
...ce page. syntax audiosourcenode = audiocontext.createmediastreamsource(stream); parameters stream a mediastream to serve as an audio source to be fed into an audio processing graph for use and manipulation. return value a new mediastreamaudiosourcenode object representing the audio node whose media is obtained from the specified source stream. 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 th...
...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 elemen...
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BaseAudioContext.createBufferSource() - Web APIs
the createbuffersource() method of the baseaudiocontext interface is used to create a new audiobuffersourcenode, which can be used to play audio data contained within an audiobuffer object. audiobuffers are created using baseaudiocontext.createbuffer or returned by baseaudiocontext.decodeaudiodata when it successfully decodes an audio track. syntax var source = baseaudiocontext.createbuffersource(); returns an audiobuffersourcenode. example in this example, we create a two second buffer, fill it with white noise, and then play it via an audiobuffersourcenode. the comments should clearly explain what is going on. note: you can also run the code live, or view the source. ...
...odata when it successfully decodes an audio track. syntax var source = baseaudiocontext.createbuffersource(); returns an audiobuffersourcenode. example in this example, we create a two second buffer, fill it with white noise, and then play it via an audiobuffersourcenode. the comments should clearly explain what is going on. note: you can also run the code live, or view the source. ...
...lly decodes an audio track. syntax var source = baseaudiocontext.createbuffersource(); returns an audiobuffersourcenode. example in this example, we create a two second buffer, fill it with white noise, and then play it via an audiobuffersourcenode. the comments should clearly explain what is going on. note: you can also run the code live, or view the source. ...
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BaseAudioContext.createConvolver() - Web APIs
the example below uses a short sample of a concert hall crowd, so the reverb effect applied is really deep and echoey. ...
...the example below uses a short sample of a concert hall crowd, so the reverb effect applied is really deep and echoey. ...
...the example below uses a short sample of a concert hall crowd, so the reverb effect applied is really deep and echoey. ...
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MediaStreamAudioSourceNode() - Web APIs
the web audio api's mediastreamaudiosourcenode() constructor creates and returns a new mediastreamaudiosourcenode object which uses the first audio track of a given mediastream as its source. note: another way to create a mediastreamaudiosourcenode is to call theaudiocontext.createmediastreamsource() method, specifying the stream from which you want to obtain audio. syntax audiosourcenode = new mediastreamaudiosourcenode(context, options); parameters context an audiocontext representing the audio context you want the node to be associated with. options a mediastreamaudiosourceoptions object defining the properties you want the mediastreamaudiosourcenode to have: mediastream a required property which specifies the mediastream from which to obtain audio for the node. retu...
...ontext, options); parameters context an audiocontext representing the audio context you want the node to be associated with. options a mediastreamaudiosourceoptions object defining the properties you want the mediastreamaudiosourcenode to have: mediastream a required property which specifies the mediastream from which to obtain audio for the node. return value a new mediastreamaudiosourcenode object representing the audio node whose media is obtained from the specified source stream. exceptions invalidstateerror the specified mediastream doesn't have any audio tracks. examples this example uses getusermedia() to obtain access to the user's camera, then creates a new mediastreamaudiosourcenode from its mediastream. ...
...alue a new mediastreamaudiosourcenode object representing the audio node whose media is obtained from the specified source stream. exceptions invalidstateerror the specified mediastream doesn't have any audio tracks. examples this example uses getusermedia() to obtain access to the user's camera, then creates a new mediastreamaudiosourcenode from its mediastream. ...
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MediaStreamAudioSourceNode.mediaStream - Web APIs
however, there is no way to be certain which track that will be on multi-track streams. ...
...however, there is no way to be certain which track that will be on multi-track streams. ...
...however, there is no way to be certain which track that will be on multi-track streams. ...
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MediaStreamTrackAudioSourceOptions.mediaStreamTrack - Web APIs
the mediastreamtrackaudiosourceoptions dictionary's mediastreamtrack property must contain a reference to the mediastreamtrack from which the mediastreamtrackaudiosourcenode being created using the mediastreamtrackaudiosourcenode() constructor. syntax mediastreamtrackaudiosourceoptions = { mediastreamtrack: audiosourcetrack; } mediastreamtrackaudiosourceoptions.mediastreamtrack = audiosourcetrack; value a mediastreamtrack from which the audio output of the new mediastreamtrackaudiosourcenode will be taken. example this example uses getusermedia() to obtain access to the user's camera, then creates a new mediastreamaudiosourcenode from the first audio track provided by the device. ...
...eamtrack: audiosourcetrack; } mediastreamtrackaudiosourceoptions.mediastreamtrack = audiosourcetrack; value a mediastreamtrack from which the audio output of the new mediastreamtrackaudiosourcenode will be taken. example this example uses getusermedia() to obtain access to the user's camera, then creates a new mediastreamaudiosourcenode from the first audio track provided by the device. ...
...ack; } mediastreamtrackaudiosourceoptions.mediastreamtrack = audiosourcetrack; value a mediastreamtrack from which the audio output of the new mediastreamtrackaudiosourcenode will be taken. example this example uses getusermedia() to obtain access to the user's camera, then creates a new mediastreamaudiosourcenode from the first audio track provided by the device. ...
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AudioBufferSourceNode.detune - Web APIs
the detune property of the audiobuffersourcenode interface is a k-rate audioparam representing detuning of oscillation in cents. for example, values of +100 and -100 detune the source up or down by one semitone, while +1200 and -1200 detune it up or down by one octave. syntax var source = audioctx.createbuffersource(); source.detune.value = 100; // value in cents note: though the audioparam returned is read-only, the value it represents is not. value a k-rate audioparam whose value indicates the detuning of oscillation in cents. ...
...nd -100 detune the source up or down by one semitone, while +1200 and -1200 detune it up or down by one octave. syntax var source = audioctx.createbuffersource(); source.detune.value = 100; // value in cents note: though the audioparam returned is read-only, the value it represents is not. value a k-rate audioparam whose value indicates the detuning of oscillation in cents. ...
...the source up or down by one semitone, while +1200 and -1200 detune it up or down by one octave. syntax var source = audioctx.createbuffersource(); source.detune.value = 100; // value in cents note: though the audioparam returned is read-only, the value it represents is not. value a k-rate audioparam whose value indicates the detuning of oscillation in cents. ...
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AudioBufferSourceNode.playbackRate - Web APIs
the playbackrate property of the audiobuffersourcenode interface is a k-rate audioparam that defines the speed at which the audio asset will be played. a value of 1.0 indicates it should play at the same speed as its sampling rate, values less than 1.0 cause the sound to play more slowly, while values greater than 1.0 result in audio playing faster than normal. the default value is 1.0. when set to another value, the audiobuffersourcenode resamples the audio before sending it to the output. syntax audiobuffersourcenode.playbackrate.value = playbackrateproportion; value an audioparam whose value is a floating-point value indicating the playback rate of the audio as a decimal proportion of the original sampling rate. ...
... default value is 1.0. when set to another value, the audiobuffersourcenode resamples the audio before sending it to the output. syntax audiobuffersourcenode.playbackrate.value = playbackrateproportion; value an audioparam whose value is a floating-point value indicating the playback rate of the audio as a decimal proportion of the original sampling rate. consider a sound buffer containing audio sampled at 44.1 khz (44,100 samples per second). let's see what a few values of playbackrate do: a playbackrate of 1.0 plays the audio at full speed, or 44,100 hz. a playbackrate of 0.5 plays the audio at half speed, or 22,050 hz. a playbackrate of 2.0 doubles the audio's playback rate to 88,200 hz. ...
...inal sampling rate. consider a sound buffer containing audio sampled at 44.1 khz (44,100 samples per second). let's see what a few values of playbackrate do: a playbackrate of 1.0 plays the audio at full speed, or 44,100 hz. a playbackrate of 0.5 plays the audio at half speed, or 22,050 hz. a playbackrate of 2.0 doubles the audio's playback rate to 88,200 hz. example in this example, the audiocontext.decodeaudiodata() function is used to decode an audio track, and put it into an audiobuffersourcenode. buttons are provided to play and stop the audio playback, and a slider control is used to change the playbackrate property value on the fly. you can run the example live (or view the source.) play the song and alter the playback rate for some fun results. <input class="playback-rate-c...
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AudioContext.getOutputTimestamp() - Web APIs
the getoutputtimestamp() property of the audiocontext interface returns a new audiotimestamp object containing two audio timestamp values relating to the current audio context. the two values are as follows: audiotimestamp.contexttime: the time of the sample frame currently being rendered by the audio output device (i.e., output audio stream position), in the same units and origin as the context's audiocontext.currenttime. basically, this is the time after the audio context was first created. audiotimestamp.performancetime: an estimation of the moment when the sample frame corresponding to the stored contexttime value was rendered by the audio output device, in the same units and origin as performance.now(). ...
...ts and origin as the context's audiocontext.currenttime. basically, this is the time after the audio context was first created. audiotimestamp.performancetime: an estimation of the moment when the sample frame corresponding to the stored contexttime value was rendered by the audio output device, in the same units and origin as performance.now(). this is the time after the document containing the audio context was first rendered. syntax var timestamp = audiocontext.getoutputtimestamp() parameters none. returns an audiotimestamp object, which has the following properties. contexttime: a point in the time coordinate system of the currenttime for the baseaudiocontext; the time after the audio context was first created. performancetime: a point in the time coordinate system of a performa...
...iocontext.getoutputtimestamp() parameters none. returns an audiotimestamp object, which has the following properties. contexttime: a point in the time coordinate system of the currenttime for the baseaudiocontext; the time after the audio context was first created. performancetime: a point in the time coordinate system of a performance interface; the time after the document containing the audio context was first rendered examples in the following code we start to play an audio file after a play button is clicked, and start off a requestanimationframe loop running, which constantly outputs the contexttime and performancetime. you can see full code of this example at output-timestamp (see it live also). ...
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AudioParam.value - Web APIs
this value will be between the values specified by the minvalue and maxvalue properties. usage notes value precision and variation the data type used internally to store value is a single-precision (32-bit) floating point number, while javascript uses 64-bit double-precision floating point numbers. ...
...this value will be between the values specified by the minvalue and maxvalue properties. usage notes value precision and variation the data type used internally to store value is a single-precision (32-bit) floating point number, while javascript uses 64-bit double-precision floating point numbers. ...
...this value will be between the values specified by the minvalue and maxvalue properties. usage notes value precision and variation the data type used internally to store value is a single-precision (32-bit) floating point number, while javascript uses 64-bit double-precision floating point numbers. ...
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Web Audio Editor - Firefox Developer Tools
notice: this tool has been deprecated and will soon be removed from firefox. for details, see deprecated tools. with the web audio api, developers create an audio context. within that context they then construct a number of audio nodes, including: nodes providing the audio source, such as an oscillator or a data buffer source nodes performing transformations such as delay and gain nodes representing the destination of the audio stream, such as the speakers each node has zero or more audioparam properties that configure its operation. for example, the gainnode has a single gain property, while the oscillatornode has frequency and detune properties. ...
...ions such as delay and gain nodes representing the destination of the audio stream, such as the speakers each node has zero or more audioparam properties that configure its operation. for example, the gainnode has a single gain property, while the oscillatornode has frequency and detune properties. the developer connects the nodes in a graph, and the complete graph defines the behavior of the audio stream. the web audio editor examines an audio context constructed in the page and provides a visualization of its graph. this gives you a high-level view of its operation, and enables you to ensure that all the nodes are connected in the way you expect. ...
... this tool is still experimental. if you find bugs, we'd love it if you filed them in bugzilla. ...
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AudioBuffer.duration - Web APIs
the duration property of the audiobuffer interface returns a double representing the duration, in seconds, of the pcm data stored in the buffer. syntax var myarraybuffer = audioctx.createbuffer(2, framecount, audioctx.samplerate); myarraybuffer.duration; value a double. ...
... the duration property of the audiobuffer interface returns a double representing the duration, in seconds, of the pcm data stored in the buffer. syntax var myarraybuffer = audioctx.createbuffer(2, framecount, audioctx.samplerate); myarraybuffer.duration; value a double. ...
...rty of the audiobuffer interface returns a double representing the duration, in seconds, of the pcm data stored in the buffer. syntax var myarraybuffer = audioctx.createbuffer(2, framecount, audioctx.samplerate); myarraybuffer.duration; value a double. ...
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AudioBuffer.length - Web APIs
the length property of the audiobuffer interface returns an integer representing the length, in sample-frames, of the pcm data stored in the buffer. syntax var myarraybuffer = audioctx.createbuffer(2, framecount, audioctx.samplerate); myarraybuffer.length; value an integer. ...
... the length property of the audiobuffer interface returns an integer representing the length, in sample-frames, of the pcm data stored in the buffer. syntax var myarraybuffer = audioctx.createbuffer(2, framecount, audioctx.samplerate); myarraybuffer.length; value an integer. ...
...f the audiobuffer interface returns an integer representing the length, in sample-frames, of the pcm data stored in the buffer. syntax var myarraybuffer = audioctx.createbuffer(2, framecount, audioctx.samplerate); myarraybuffer.length; value an integer. ...
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AudioBuffer.numberOfChannels - Web APIs
the numberofchannels property of the audiobuffer interface returns an integer representing the number of discrete audio channels described by the pcm data stored in the buffer. syntax var myarraybuffer = audioctx.createbuffer(2, framecount, audioctx.samplerate); myarraybuffer.numberofchannels; value an integer. ...
...of the audiobuffer interface returns an integer representing the number of discrete audio channels described by the pcm data stored in the buffer. syntax var myarraybuffer = audioctx.createbuffer(2, framecount, audioctx.samplerate); myarraybuffer.numberofchannels; value an integer. ...
...eturns an integer representing the number of discrete audio channels described by the pcm data stored in the buffer. syntax var myarraybuffer = audioctx.createbuffer(2, framecount, audioctx.samplerate); myarraybuffer.numberofchannels; value an integer. ...
...And 4 more matches

AudioBuffer.sampleRate - Web APIs
the samplerate property of the audiobuffer interface returns a float representing the sample rate, in samples per second, of the pcm data stored in the buffer. syntax var myarraybuffer = audioctx.createbuffer(2, framecount, audioctx.samplerate); myarraybuffer.samplerate; value a floating-point value indicating the current sample rate of the buffers data, in samples per second. ...
..., in samples per second, of the pcm data stored in the buffer. syntax var myarraybuffer = audioctx.createbuffer(2, framecount, audioctx.samplerate); myarraybuffer.samplerate; value a floating-point value indicating the current sample rate of the buffers data, in samples per second. ...
...e pcm data stored in the buffer. syntax var myarraybuffer = audioctx.createbuffer(2, framecount, audioctx.samplerate); myarraybuffer.samplerate; value a floating-point value indicating the current sample rate of the buffers data, in samples per second. ...
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AudioContext() - Web APIs
the audiocontext() constructor creates a new audiocontext object which represents an audio-processing graph, built from audio modules linked together, each represented by an audionode. syntax var audioctx = new audiocontext(); var audioctx = new audiocontext(options); parameters options optional an object based on the audiocontextoptions dictionary that contains zero or more optional properties to configure the new context. available properties are as follows: latencyhint optional the type of playback that the context will be used for, as a value from the audiocontextlatencycategory enum or a double-precision floating-point value indicating the preferred maximum latency of the context in seconds. ...
...properties to configure the new context. available properties are as follows: latencyhint optional the type of playback that the context will be used for, as a value from the audiocontextlatencycategory enum or a double-precision floating-point value indicating the preferred maximum latency of the context in seconds. the user agent may or may not choose to meet this request; check the value of audiocontext.baselatency to determine the true latency after creating the context. samplerate optional the samplerate to be used by the audiocontext, specified in samples per second. the value may be any value supported by audiobuffer. if not specified, the preferred sample rate for the context's output device is used by default. return value the newly constructed audiocontext instance. exceptio...
...cond. the value may be any value supported by audiobuffer. if not specified, the preferred sample rate for the context's output device is used by default. return value the newly constructed audiocontext instance. exceptions notsupportederror the specified samplerate isn't supported by the context. usage notes the specification doesn't go into a lot of detail about things like how many audio contexts a user agent should support, or minimum or maximum latency requirements (if any), so these details can vary from browser to browser. be sure to check the values if they matter to you. in particular, the specification doesn't indicate a maximum or minimum number of audio contexts that must be able to be open at the same time, so this is left up to the browser implementations to decide. ...
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AudioContext.createMediaStreamDestination() - Web APIs
the createmediastreamdestination() method of the audiocontext interface is used to create a new mediastreamaudiodestinationnode object associated with a webrtc mediastream representing an audio stream, which may be stored in a local file or sent to another computer. the mediastream is created when the node is created and is accessible via the mediastreamaudiodestinationnode's stream attribute. this stream can be used in a similar way as a mediastream obtained via navigator.getusermedia — it can, for example, be sent to a remote peer using the rtcpeerconnection addstream() method. for more details about media stream destination nodes, check out the mediastreamaudiodestinationnode reference page. ...
...nd is accessible via the mediastreamaudiodestinationnode's stream attribute. this stream can be used in a similar way as a mediastream obtained via navigator.getusermedia — it can, for example, be sent to a remote peer using the rtcpeerconnection addstream() method. for more details about media stream destination nodes, check out the mediastreamaudiodestinationnode reference page. syntax var audioctx = new audiocontext(); var destination = audioctx.createmediastreamdestination(); returns a mediastreamaudiodestinationnode. ...
...t to a remote peer using the rtcpeerconnection addstream() method. for more details about media stream destination nodes, check out the mediastreamaudiodestinationnode reference page. syntax var audioctx = new audiocontext(); var destination = audioctx.createmediastreamdestination(); returns a mediastreamaudiodestinationnode. examples in the following simple example, we create a mediastreamaudiodestinationnode, an oscillatornode and a mediarecorder (the example will therefore only work in firefox and chrome at this time.) the mediarecorder is set up to record information from the mediastreamdestinationnode. when the button is clicked, the oscillator starts, and the mediarecorder is started. when the button is stopped, the oscillator and mediarecorder both stop. stopping the mediarecorde...
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AudioNode.channelInterpretation - Web APIs
the m input channel is used for both output channels (l and r). ...
... the m input channel is used for both output channels (l and r). ...
... the m input channel is used for both output channels (l and r). ...
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AudioParam.cancelAndHoldAtTime() - Web APIs
working draft initial definition. ...
... working draft initial definition. ...
... working draft initial definition. ...
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AudioTrack.kind - Web APIs
the kind property contains a string indicating the category of audio contained in the audiotrack. the kind can be used to determine the scenarios in which specific tracks should be enabled or disabled. see audio track kind strings for a list of the kinds available for audio tracks. syntax var trackkind = audiotrack.kind; value a domstring specifying the type of content the media represents. the string is one of those found in audio track kind strings below. audio track kind strings the kinds available for audio tracks are: "alternative" a potential alternative to the main track, such as a different audio take or a version of the soundtrack with only the music and no dialogue. "descriptions" an audio track providing audible descriptions of the action depicted in a video track. "mai...
...this might be used to contain the director's commentary track on a movie, for example. ...
...this might be used to contain the director's commentary track on a movie, for example. ...
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AudioTrack.label - Web APIs
the list is filtered to only allow certain track kinds through. ...
...the list is filtered to only allow certain track kinds through. ...
...the list is filtered to only allow certain track kinds through. ...
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AudioTrack.language - Web APIs
for example, if the primary language used in the track is united states english, this value would be "en-us". for brazilian portuguese, the value would be "pt-br". ...
... for example, if the primary language used in the track is united states english, this value would be "en-us". for brazilian portuguese, the value would be "pt-br". ...
... for example, if the primary language used in the track is united states english, this value would be "en-us". for brazilian portuguese, the value would be "pt-br". ...
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AudioWorkletGlobalScope - Web APIs
the audioworkletglobalscope interface of the web audio api represents a global execution context for user-supplied code, which defines custom audioworkletprocessor-derived classes. each baseaudiocontext has a single audioworklet available under the audioworklet property, which runs its code in a single audioworkletglobalscope. as the global execution context is shared across the current baseaudiocontext, it's possible to define any other variables and perform any actions allowed in worklets — apart from defining audioworkletprocessor-derived classes. properties currentframe read only returns an integer that represents the ever-increasing current sample-frame of the audio block being processed. ...
...diocontext, it's possible to define any other variables and perform any actions allowed in worklets — apart from defining audioworkletprocessor-derived classes. properties currentframe read only returns an integer that represents the ever-increasing current sample-frame of the audio block being processed. it is incremented by 128 (the size of a render quantum) after the processing of each audio block. currenttime read only returns a double that represents the ever-increasing context time of the audio block being processed. it is equal to the currenttime property of the baseaudiocontext the worklet belongs to. samplerate read only returns a float that represents the sample rate of the associated baseaudiocontext. methods registerprocessor() registers a class derived from the ...
... block. currenttime read only returns a double that represents the ever-increasing context time of the audio block being processed. it is equal to the currenttime property of the baseaudiocontext the worklet belongs to. samplerate read only returns a float that represents the sample rate of the associated baseaudiocontext. methods registerprocessor() registers a class derived from the audioworkletprocessor interface. the class can then be used by creating an audioworkletnode, providing its registered name. examples in this example we output all global properties into the console in the constructor of a custom audioworkletprocessor. first we need to define the processor, and register it. note that this should be done in a separate file. // test-processor.js class testprocessor e...
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AudioWorkletNode - Web APIs
although the interface is available outside secure contexts, the baseaudiocontext.audioworklet property is not, thus custom audioworkletprocessors cannot be defined outside them. the audioworkletnode interface of the web audio api represents a base class for a user-defined audionode, which can be connected to an audio routing graph along with other nodes. it has an associated audioworkletprocessor, which does the actual audio processing in a web audio rendering thread. constructor audioworkletnode() creates a new instance of an audioworkletnode object. properties also inherits properties from its parent, audionode. audioworkletnode.port read only returns a messageport used for bidirectional communication between the node and its associated audioworkletprocessor. the other end is ...
...rendering thread. constructor audioworkletnode() creates a new instance of an audioworkletnode object. properties also inherits properties from its parent, audionode. audioworkletnode.port read only returns a messageport used for bidirectional communication between the node and its associated audioworkletprocessor. the other end is available under the port property of the processor. audioworkletnode.parameters read only returns an audioparammap — a collection of audioparam objects. they are instantiated during the creation of the underlying audioworkletprocessor. if the audioworkletprocessor has a static parameterdescriptors getter, the audioparamdescriptor array returned from it is used to create audioparam objects on the audioworkletnode. with this mechanism it is possible t...
...meters read only returns an audioparammap — a collection of audioparam objects. they are instantiated during the creation of the underlying audioworkletprocessor. if the audioworkletprocessor has a static parameterdescriptors getter, the audioparamdescriptor array returned from it is used to create audioparam objects on the audioworkletnode. with this mechanism it is possible to make your own audioparam objects accessible from your audioworkletnode. you can then use their values in the associated audioworkletprocessor. event handlers audioworkletnode.onprocessorerror fired when an error is thrown in associated audioworkletprocessor. once fired, the processor and consequently the node will output silence throughout its lifetime. ...
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BaseAudioContext.createGain() - Web APIs
the creategain() method of the baseaudiocontext interface creates a gainnode, which can be used to control the overall gain (or volume) of the audio graph. syntax var gainnode = audiocontext.creategain(); return value a gainnode which takes as input one or more audio sources and outputs audio whose volume has been adjusted in gain (volume) to a level specified by the node's gainnode.gain a-rate parameter. example the following example shows basic usage of an audiocontext to create a gainnode, which is then used to mute and unmute the audio when a mute button is clicked by changing the gain property value. the below snippet wouldn't work as is — for a complete working example, check out our voice-change-o-matic demo (view source.) <div> <button class="mute">mute button</button> </...
... example the following example shows basic usage of an audiocontext to create a gainnode, which is then used to mute and unmute the audio when a mute button is clicked by changing the gain property value. ...
...g example shows basic usage of an audiocontext to create a gainnode, which is then used to mute and unmute the audio when a mute button is clicked by changing the gain property value. ...
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OfflineAudioContext: complete event - Web APIs
the complete event of the offlineaudiocontext interface is fired when the rendering of an offline audio context is complete. ...
... the complete event of the offlineaudiocontext interface is fired when the rendering of an offline audio context is complete. ...
... the complete event of the offlineaudiocontext interface is fired when the rendering of an offline audio context is complete. ...
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Live streaming web audio and video - Developer guides
onitored and if it looks like it's not keeping up, we drop down to a lower bandwidth (and consequently lower quality) stream. in order to have this capability, we need to use formats that facilitate this. live streaming formats generally allow adaptive streaming by breaking streams into a series of small segments and making those segments available at different qualities and bit rates. ...
...o use formats that facilitate this. live streaming formats generally allow adaptive streaming by breaking streams into a series of small segments and making those segments available at different qualities and bit rates. streaming audio and video on demand streaming technology is not used exclusively for live streams. ...
...ivery. using rtp with rtcp allows for adaptive streaming. this is not yet supported natively in most browsers, but be aware that firefox os 1.3 supports rtsp. note: some vendors implement propriety transport protocols, such as realnetworks and their real data transport (rdt). rtsp 2.0 rtsp 2.0 is currently in development and is not backward compatible with rtsp 1.0. important: although the <audio> and <video> tags are protocol agnostic, no browser currently supports anything other than http without requiring plugins, although this looks set to change. protocols other than http may also be subject to blocking from firewalls or proxy servers. using streaming protocols the process of using the various protocols is reassuringly familiar if you are used to working with media over http. for...
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Visualizing an audio spectrum - Archive of obsolete content
this example calculates and displays fast fourier transform (fft) spectrum data for the playing audio. the function handling the loadedmetadata event stores the metadata of the audio element in global variables; the function for the mozaudioavailable event does an fft of the samples and displays them in a canvas. note: you can use the audionode called analysernode to perform real-time fft analysis on an audio stream, rather than the code shown below. ...
...tadata of the audio element in global variables; the function for the mozaudioavailable event does an fft of the samples and displays them in a canvas. note: you can use the audionode called analysernode to perform real-time fft analysis on an audio stream, rather than the code shown below. ...
...the audio element in global variables; the function for the mozaudioavailable event does an fft of the samples and displays them in a canvas. note: you can use the audionode called analysernode to perform real-time fft analysis on an audio stream, rather than the code shown below. ...
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Displaying a graphic with audio samples - Archive of obsolete content
the following example shows how to take samples from an audio stream and display them behind an image (in this case, the mozilla logo), giving the impression that the image is built from the samples. ...
... the following example shows how to take samples from an audio stream and display them behind an image (in this case, the mozilla logo), giving the impression that the image is built from the samples. ...
... the following example shows how to take samples from an audio stream and display them behind an image (in this case, the mozilla logo), giving the impression that the image is built from the samples. ...
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Video and Audio APIs - Learn web development
previous overview: client-side web apis next html5 comes with elements for embedding rich media in documents — <video> and <audio> — which in turn come with their own apis for controlling playback, seeking, etc. this article shows you how to do common tasks such as creating custom playback controls. prerequisites: javascript basics (see first steps, building blocks, javascript objects), the basics of client-side apis objective: to learn how to use browser apis to control video and audio playback. html5 video and audio the <video> and <audio> elements allow us to embed video and audio into web pages. ...
...creating custom playback controls. prerequisites: javascript basics (see first steps, building blocks, javascript objects), the basics of client-side apis objective: to learn how to use browser apis to control video and audio playback. html5 video and audio the <video> and <audio> elements allow us to embed video and audio into web pages. as we showed in video and audio content, a typical implementation looks like this: <video controls> <source src="rabbit320.mp4" type="video/mp4"> <source src="rabbit320.webm" type="video/webm"> <p>your browser doesn't support html5 video. here is a <a href="rabbit320.mp4">link to the video</a> instead.</p> </video> this creates a video player inside the browser like so: you can review what all the html features do in...
...ren't very keyboard-accessible. you can solve both these problems by hiding the native controls (by removing the controls attribute), and programming your own with html, css, and javascript. in the next section we'll look at the basic tools we have available to do this. the htmlmediaelement api part of the html5 spec, the htmlmediaelement api provides features to allow you to control video and audio players programmatically — for example htmlmediaelement.play(), htmlmediaelement.pause(), etc. this interface is available to both <audio> and <video> elements, as the features you'll want to implement are nearly identical. let's go through an example, adding features as we go. our finished example will look (and function) something like the following: getting started to get started with t...
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AudioBuffer() - Web APIs
numberofchannels: the number of channels for the buffer. the default is 1, and all user agents are required to support at least 32 channels. samplerate: the sample rate in hz for the buffer. the default is the sample rate of the context used in constructing this object. ...
... numberofchannels: the number of channels for the buffer. the default is 1, and all user agents are required to support at least 32 channels. samplerate: the sample rate in hz for the buffer. the default is the sample rate of the context used in constructing this object. ...
...default is 1, and all user agents are required to support at least 32 channels. samplerate: the sample rate in hz for the buffer. the default is the sample rate of the context used in constructing this object. user agents are required to support sample rates from 8,000 hz to 96,000 hz (but are allowed to go farther outside this range). deprecated parameters context a reference to an audiocontext. this parameter was removed from the spec. return value a new audiobuffer object instance. exceptions notsupportederror one or more of the options are negative or otherwise has an invalid value (such as numberofchannels being higher than supported, or a samplerate outside the nominal range). rangeerror there isn't enough memory available to allocate the buffer. specifications ...
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AudioBufferSourceNode.start() - Web APIs
the start() method of the audiobuffersourcenode interface is used to schedule playback of the audio data contained in the buffer, or to begin playback immediately. syntax audiobuffersourcenode.start([when][, offset][, duration]); parameters when optional the time, in seconds, at which the sound should begin to play, in the same time coordinate system used by the audiocontext. if when is less than (audiocontext.currenttime, or if it's 0, the sound begins to play at once. the default value is 0. offset optional an offset, specified as the number of seconds in the same time coordinate system as the audiocontext, to the time within the audio buffer that playback should begin. for example, to start playback halfway through a 10-second audio clip, offset should be 5. the default value, 0...
... the sound begins to play at once. the default value is 0. offset optional an offset, specified as the number of seconds in the same time coordinate system as the audiocontext, to the time within the audio buffer that playback should begin. for example, to start playback halfway through a 10-second audio clip, offset should be 5. the default value, 0, will begin playback at the beginning of the audio buffer, and offsets past the end of the audio which will be played (based on the audio buffer's duration and/or the loopend property) are silently clamped to the maximum value allowed. the computation of the offset into the sound is performed using the sound buffer's natural sample rate, rather than the current playback rate, so even if the sound is playing at twice its normal speed, the midway p...
... duration optional the duration of the sound to be played, specified in seconds. if this parameter isn't specified, the sound plays until it reaches its natural conclusion or is stopped using the stop() method. using this parameter is functionally identical to calling start(when, offset) and then calling stop(when+duration). return value undefined. ...
...And 3 more matches

AudioNode.channelCountMode - Web APIs
in this case, channelcount is ignored and only up-mixing happens. gainnode, delaynode, scriptprocessornode, channelmergernode, biquadfilternode, waveshapernode clamped-max the number of channels is equal to the maximum number of channels of all connections, clamped to the value of channelcount. ...
...xing happens. gainnode, delaynode, scriptprocessornode, channelmergernode, biquadfilternode, waveshapernode clamped-max the number of channels is equal to the maximum number of channels of all connections, clamped to the value of channelcount. pannernode, convolvernode, dynamicscompressornode explicit the number of channels is defined by the value of channelcount. ...
...the maximum number of channels of all connections, clamped to the value of channelcount. pannernode, convolvernode, dynamicscompressornode explicit the number of channels is defined by the value of channelcount. audiodestinationnode, analysernode, channelsplitternode in older versions of the spec, the default for a channelsplitternode was max. syntax var oscillator = audioctx.createoscillator(); oscillator.channelcountmode = 'explicit'; value a enumerated value representing a channelcountmode. ...
...And 3 more matches

AudioNode.context - Web APIs
the read-only context property of the audionode interface returns the associated baseaudiocontext, that is the object representing the processing graph the node is participating in. syntax var acontext = anaudionode.context; value the audiocontext or offlineaudiocontext object that was used to construct this audionode. ...
... the read-only context property of the audionode interface returns the associated baseaudiocontext, that is the object representing the processing graph the node is participating in. syntax var acontext = anaudionode.context; value the audiocontext or offlineaudiocontext object that was used to construct this audionode. ...
... the read-only context property of the audionode interface returns the associated baseaudiocontext, that is the object representing the processing graph the node is participating in. syntax var acontext = anaudionode.context; value the audiocontext or offlineaudiocontext object that was used to construct this audionode. ...
...And 3 more matches

AudioNode.disconnect() - Web APIs
the disconnect() method of the audionode interface lets you disconnect one or more nodes from the node on which the method is called. syntax audionode.disconnect(); audionode.disconnect(output); audionode.disconnect(destination); audionode.disconnect(destination, output); audionode.disconnect(destination, output, input); return value undefined parameters there are several versions of the disconnect() method, which accept different combinations of parameters to control which nodes to disconnect from. if no parameters are provided, all outgoing connections are disconnected. destination optional an audionode or audioparam specifying the node or nodes to disconnect from. ...
..., output, input); return value undefined parameters there are several versions of the disconnect() method, which accept different combinations of parameters to control which nodes to disconnect from. if no parameters are provided, all outgoing connections are disconnected. destination optional an audionode or audioparam specifying the node or nodes to disconnect from. if this value is an audionode, a single node is disconnected from, with any other, optional, parameters (output and/or input) further limiting which inputs and/or outputs should be disconnected. if this value is an audioparam, then the connection to that audioparam is terminated, and the node's contributions to that computed parameter become 0 going forward once the change takes effect. if no matching connection is found,...
...iting which inputs and/or outputs should be disconnected. if this value is an audioparam, then the connection to that audioparam is terminated, and the node's contributions to that computed parameter become 0 going forward once the change takes effect. if no matching connection is found, an invalidaccesserror exception is thrown. output optional an index describing which output from the current audionode is to be disconnected. the index numbers are defined according to the number of output channels (see audio channels). if this parameter is out-of-bound, an indexsizeerror exception is thrown. input optional an index describing which input into the specified destination audionode is to be disconnected. ...
...And 3 more matches

AudioScheduledSourceNode.start() - Web APIs
if 0 is passed then the playback will start from the beginning. ...
...if 0 is passed then the playback will start from the beginning. ...
...if 0 is passed then the playback will start from the beginning. ...
...And 3 more matches

AudioScheduledSourceNode - Web APIs
silence is represented, as always, by a stream of samples with the value zero (0). ...
...silence is represented, as always, by a stream of samples with the value zero (0). ...
...silence is represented, as always, by a stream of samples with the value zero (0). ...
...And 3 more matches

AudioTrackList.getTrackById() - Web APIs
if no match is found, this method returns null. the tracks are searched in their natural order; that is, in the order defined by the media resource itself, or, if the resource doesn't define an order, the relative order in which the tracks are declared by the media resource. ...
...if no match is found, this method returns null. the tracks are searched in their natural order; that is, in the order defined by the media resource itself, or, if the resource doesn't define an order, the relative order in which the tracks are declared by the media resource. ...
...if no match is found, this method returns null. the tracks are searched in their natural order; that is, in the order defined by the media resource itself, or, if the resource doesn't define an order, the relative order in which the tracks are declared by the media resource. ...
...And 3 more matches

AudioWorkletNodeOptions - Web APIs
properties numberofinputs optional the value to initialize the numberofinputs property to. defaults to 1. numberofoutputs optional the value to initialize the numberofoutputs property to. defaults to 1. outputchannelcount optional an array defining the number of channels for each output. ...
... properties numberofinputs optional the value to initialize the numberofinputs property to. defaults to 1. numberofoutputs optional the value to initialize the numberofoutputs property to. defaults to 1. outputchannelcount optional an array defining the number of channels for each output. ...
... initialize the numberofoutputs property to. defaults to 1. outputchannelcount optional an array defining the number of channels for each output. for example, outputchannelcount: [n, m] specifies the number of channels in the first output to be n and the second output to be m. the array length must match numberofoutputs. parameterdata optional an object containing the initial values of custom audioparam objects on this node (in its parameters property), with key being the name of a custom parameter and value being its initial value. processoroptions optional any additional data that can be used for custom initialization of the underlying audioworkletprocessor. usage notes when creating an audioworkletnode, these options can have various effects. if the number of inputs and number of ou...
...And 3 more matches

BaseAudioContext.createOscillator() - Web APIs
the createoscillator() method of the baseaudiocontext interface creates an oscillatornode, a source representing a periodic waveform. it basically generates a constant tone. syntax var oscillatornode = audioctx.createoscillator(); returns an oscillatornode. example the following example shows basic usage of an audiocontext to create an oscillator node. for applied examples/information, check out our violent theremin demo (see app.js for relevant code); also see our oscillatornode page for more information. ...
... it basically generates a constant tone. syntax var oscillatornode = audioctx.createoscillator(); returns an oscillatornode. example the following example shows basic usage of an audiocontext to create an oscillator node. for applied examples/information, check out our violent theremin demo (see app.js for relevant code); also see our oscillatornode page for more information. ...
...s a constant tone. syntax var oscillatornode = audioctx.createoscillator(); returns an oscillatornode. example the following example shows basic usage of an audiocontext to create an oscillator node. for applied examples/information, check out our violent theremin demo (see app.js for relevant code); also see our oscillatornode page for more information. ...
...And 3 more matches

AudioContextOptions - Web APIs
the audiocontextoptions dictionary is used to specify configuration options when constructing a new audiocontext object to represent a graph of web audio nodes. it is only used when calling the audiocontext() constructor. properties latencyhint optional the type of playback that the context will be used for, as a value from the audiocontextlatencycategory enum or a double-precision floating-point value indicating the preferred maximum latency of the context in seconds. the user agent may or may not choose to meet this request; check the value of audiocontext.baselatency to determine the true latency after creating the context. samplerate optional the samplerate to be used by the audiocontext, specified in samples per second. the value may be any value supported by audiobuffer. if not sp...
...rate optional the samplerate to be used by the audiocontext, specified in samples per second. the value may be any value supported by audiobuffer. if not specified, the preferred sample rate for the context's output device is used by default. constants standard values for latencyhint the latencyhint property can be number specifying a preferred maximum latency in seconds or a string from the audiocontextlatencycategory enumerated string, which selects a standard value for a given type of audio usage: value description "balanced" the user agent should balance audio output latency and power consumption when selecting a latency value. ...
...this is likely to require increased power consumption. ...
...And 2 more matches

AudioWorkletNode() - Web APIs
the audioworkletnode() constructor creates a new audioworkletnode object, which represents an audionode that uses a javascript function to perform custom audio processing. syntax var node = new audioworkletnode(context, name); var node = new audioworkletnode(context, name, options); parameters context the baseaudiocontext instance this node will be associated with. name a string, which represents the name of the audioworkletprocessor this node will be based on. a processor with the provided name must first be registered using the audioworkletglobalscope.registerprocessor() method. options optional an object based on the audioworkletnodeoptions dictionary, which contains zero or more optional properties to configure the new node. the result of the structured clone algorithm applied ...
...available properties are as follows: numberofinputs optional the value to initialize the numberofinputs property to. defaults to 1. ...
...available properties are as follows: numberofinputs optional the value to initialize the numberofinputs property to. defaults to 1. numberofoutputs optional the value to initialize the numberofoutputs property to. defaults to 1. ...
...And 2 more matches

BaseAudioContext.createChannelMerger() - Web APIs
the createchannelmerger() method of the baseaudiocontext interface creates a channelmergernode, which combines channels from multiple audio streams into a single audio stream. syntax baseaudiocontext.createchannelmerger(numberofinputs); parameters numberofinputs the number of channels in the input audio streams, which the output stream will contain; the default is 6 if this parameter is not specified. returns a channelmergernode. example the following example shows how you could separate a stereo track (say, a piece of music), and process the left and right channel differently. ...
...uts the number of channels in the input audio streams, which the output stream will contain; the default is 6 if this parameter is not specified. returns a channelmergernode. example the following example shows how you could separate a stereo track (say, a piece of music), and process the left and right channel differently. to use them, you need to use the second and third parameters of the audionode.connect(audionode) method, which allow you to specify both the index of the channel to connect from and the index of the channel to connect to. ...
...ergernode. example the following example shows how you could separate a stereo track (say, a piece of music), and process the left and right channel differently. to use them, you need to use the second and third parameters of the audionode.connect(audionode) method, which allow you to specify both the index of the channel to connect from and the index of the channel to connect to. ...
...And 2 more matches

BaseAudioContext.createChannelSplitter() - Web APIs
the createchannelsplitter() method of the baseaudiocontext interface is used to create a channelsplitternode, which is used to access the individual channels of an audio stream and process them separately. syntax baseaudiocontext.createchannelsplitter(numberofoutputs); parameters numberofoutputs the number of channels in the input audio stream that you want to output separately; the default is 6 if this parameter is not specified. returns a channelsplitternode. example the following simple example shows how you could separate a stereo track (say, a piece of music), and process the left and right channel differently. ...
...the number of channels in the input audio stream that you want to output separately; the default is 6 if this parameter is not specified. returns a channelsplitternode. example the following simple example shows how you could separate a stereo track (say, a piece of music), and process the left and right channel differently. to use them, you need to use the second and third parameters of the audionode.connect(audionode) method, which allow you to specify the index of the channel to connect from and the index of the channel to connect to. ...
...ternode. example the following simple example shows how you could separate a stereo track (say, a piece of music), and process the left and right channel differently. to use them, you need to use the second and third parameters of the audionode.connect(audionode) method, which allow you to specify the index of the channel to connect from and the index of the channel to connect to. ...
...And 2 more matches

DisplayMediaStreamConstraints.audio - Web APIs
the displaymediastreamconstraints dictionary's audio property is used to specify whether or not to request that the mediastream containing screen display contents also include an audio track. this value may simply be a boolean, where true indicates that an audio track should be included an false (the default) indicates that no audio should be included in the stream. more precise control over the audio data may be exercised by instead providing a mediatrackconstraints object, which is used to process the audio prior to adding it to the stream. note: the specification for the screen capture api does not define what the contents of the audio track should be. this is left up to the user agent to decide. ...
... in the stream. more precise control over the audio data may be exercised by instead providing a mediatrackconstraints object, which is used to process the audio prior to adding it to the stream. note: the specification for the screen capture api does not define what the contents of the audio track should be. this is left up to the user agent to decide. likely sources are the computer's system audio output and the user's microphone, but there is no way to specify which source to use. syntax displaymediastreamconstraints.audio = allowaudioflag; displaymediastreamconstraints.audio = mediatrackconstraints; displaymediastreamconstraints = { audio: allowaudioflag|mediatrackconstraints; } value the value may be either a boolean or a mediatrackconstraints object. if a boolean is specified...
...ut there is no way to specify which source to use. syntax displaymediastreamconstraints.audio = allowaudioflag; displaymediastreamconstraints.audio = mediatrackconstraints; displaymediastreamconstraints = { audio: allowaudioflag|mediatrackconstraints; } value the value may be either a boolean or a mediatrackconstraints object. if a boolean is specified, a value of true indicates that an audio track should be included in the stream returned by getdisplaymedia(), if an appropriate audio source exists and the user agent supports audio on display media. a value of false—the default—indicates that no audio track is to be included in the stream. ...
...And 2 more matches

AudioBuffer.copyToChannel() - Web APIs
the copytochannel() method of the audiobuffer interface copies the samples to the specified channel of the audiobuffer, from the source array. syntax myarraybuffer.copytochannel(source, channelnumber, startinchannel); parameters source a float32array that the channel data will be copied from. channelnumber the channel number of the current audiobuffer to copy the channel data to. if channelnumber is greater than or equal to audiobuffer.numberofchannels, an index_size_err will be thrown. startinchannel optional an optional offset to copy the data to. if startinchannel is greater than audiobuffer.length, an index_size_err will be thrown. ...
...myarraybuffer.copyfromchannel(anotherarray,1,0); // copy data from anotherarray to first channel of myarraybuffer. both channels have the same data now. ...
...annel number of the current audiobuffer to copy the channel data to. if channelnumber is greater than or equal to audiobuffer.numberofchannels, an index_size_err will be thrown. startinchannel optional an optional offset to copy the data to. if startinchannel is greater than audiobuffer.length, an index_size_err will be thrown. example var myarraybuffer = audioctx.createbuffer(2, framecount, audioctx.samplerate); var anotherarray = new float32array; // copy channel data from second channel of myarraybuffer. myarraybuffer.copyfromchannel(anotherarray,1,0); // copy data from anotherarray to first channel of myarraybuffer. both channels have the same data now. ...
...count, audioctx.samplerate); var anotherarray = new float32array; // copy channel data from second channel of myarraybuffer. myarraybuffer.copyfromchannel(anotherarray,1,0); // copy data from anotherarray to first channel of myarraybuffer. both channels have the same data now. myarraybuffer.copytochannel (anotherarray,0,0); specification specification status comment web audio apithe definition of 'copytochannel' in that specification. working draft ...

AudioBufferSourceNode.buffer - Web APIs
the buffer property of the audiobuffersourcenode interface provides the ability to play back audio using an audiobuffer as the source of the sound data. if the buffer property is set to the value null, the node generates a single channel containing silence (that is, every sample is 0). syntax audiobuffersourcenode.buffer = soundbuffer; value an audiobuffer which contains the data representing the sound which the node will play. example for a full working example, see this code running live, or view the source. ...
...data. if the buffer property is set to the value null, the node generates a single channel containing silence (that is, every sample is 0). syntax audiobuffersourcenode.buffer = soundbuffer; value an audiobuffer which contains the data representing the sound which the node will play. example for a full working example, see this code running live, or view the source. ...
...to the value null, the node generates a single channel containing silence (that is, every sample is 0). syntax audiobuffersourcenode.buffer = soundbuffer; value an audiobuffer which contains the data representing the sound which the node will play. example for a full working example, see this code running live, or view the source. ...
...x.samplerate); button.onclick = function() { // fill the buffer with white noise; //just random values between -1.0 and 1.0 for (var channel = 0; channel < channels; channel++) { // this gives us the actual arraybuffer that contains the data var nowbuffering = myarraybuffer.getchanneldata(channel); for (var i = 0; i < framecount; i++) { // math.random() is in [0; 1.0] // audio needs to be in [-1.0; 1.0] nowbuffering[i] = math.random() * 2 - 1; } } // get an audiobuffersourcenode. // this is the audionode to use when we want to play an audiobuffer var source = audioctx.createbuffersource(); // set the buffer in the audiobuffersourcenode source.buffer = myarraybuffer; specifications specification status comment web audio...

AudioContext.suspend() - Web APIs
the suspend() method of the audiocontext interface suspends the progression of time in the audio context, temporarily halting audio hardware access and reducing cpu/battery usage in the process — this is useful if you want an application to power down the audio hardware when it will not be using an audio context for a while. this method will cause an invalid_state_err exception to be thrown if called on an offlineaudiocontext. syntax var audioctx = new audiocontext(); audioctx.suspend().then(function() { ... }); returns a promise that resolves with void. the promise is rejected if the context has already been closed. ...
...ll not be using an audio context for a while. this method will cause an invalid_state_err exception to be thrown if called on an offlineaudiocontext. syntax var audioctx = new audiocontext(); audioctx.suspend().then(function() { ... }); returns a promise that resolves with void. the promise is rejected if the context has already been closed. example the following snippet is taken from our audiocontext states demo (see it running live.) when the suspend/resume button is clicked, the audiocontext.state is queried — if it is running, suspend() is called; if it is suspended, resume() is called. in each case, the text label of the button is updated as appropriate once the promise resolves. ...
...losed. example the following snippet is taken from our audiocontext states demo (see it running live.) when the suspend/resume button is clicked, the audiocontext.state is queried — if it is running, suspend() is called; if it is suspended, resume() is called. in each case, the text label of the button is updated as appropriate once the promise resolves. ...
...et is taken from our audiocontext states demo (see it running live.) when the suspend/resume button is clicked, the audiocontext.state is queried — if it is running, suspend() is called; if it is suspended, resume() is called. in each case, the text label of the button is updated as appropriate once the promise resolves. susresbtn.onclick = function() { if(audioctx.state === 'running') { audioctx.suspend().then(function() { susresbtn.textcontent = 'resume context'; }); } else if(audioctx.state === 'suspended') { audioctx.resume().then(function() { susresbtn.textcontent = 'suspend context'; }); } } specifications specification status comment web audio apithe definition of 'close()' in that specification. working draft ...

AudioContextLatencyCategory - Web APIs
the audiocontextlatencycategory type is an enumerated set of strings which are used to select one of a number of default values for acceptable maximum latency of an audio context. by using these strings rather than a numeric value when specifying a latency to a audiocontext, you can allow the user agent to select an appropriate latency for your use case that makes sense on the device on which your content is being used. audiocontextlatencycategory can be used when constructing a new audiocontext by passing one of these values as the latencyhint option in the audiocontext() constructor's options dictionary. values value description "balanced" the user agent should balance audio output latency and power consumption when selecting a latency value. "in...
...this is likely to require increased power consumption. this is the default value. ...
... "balanced" the user agent should balance audio output latency and power consumption when selecting a latency value. "interactive" the audio is involved in interactive elements, such as responding to user actions or needing to coincide with visual cues such as a video or game action. the user agent should select the lowest possible latency that doesn't cause glitches in the audio. this is likely to require increased power consumption. this is the default value. "playback" the user agent should select a latency that will maximize playback time by minimizing power consumption at the expense of latency. useful for non-interactive playback, such as playing music. ...
... glitches in the audio. this is likely to require increased power consumption. this is the default value. "playback" the user agent should select a latency that will maximize playback time by minimizing power consumption at the expense of latency. useful for non-interactive playback, such as playing music. specifications specification status comment web audio apithe definition of 'audiocontextlatencycategory' in that specification. working draft initial definition. ...

AudioDestinationNode.maxChannelCount - Web APIs
the maxchannelcount property of the audiodestinationnode interface is an unsigned long defining the maximum amount of channels that the physical device can handle. the audionode.channelcount property can be set between 0 and this value (both included). if maxchannelcount is 0, like in offlineaudiocontext, the channel count cannot be changed. syntax var audioctx = new audiocontext(); var mydestination = audioctx.destination; mydestination.maxchannelcount = 2; value an unsigned long. ...
...ical device can handle. the audionode.channelcount property can be set between 0 and this value (both included). if maxchannelcount is 0, like in offlineaudiocontext, the channel count cannot be changed. syntax var audioctx = new audiocontext(); var mydestination = audioctx.destination; mydestination.maxchannelcount = 2; value an unsigned long. ...
...he audionode.channelcount property can be set between 0 and this value (both included). if maxchannelcount is 0, like in offlineaudiocontext, the channel count cannot be changed. syntax var audioctx = new audiocontext(); var mydestination = audioctx.destination; mydestination.maxchannelcount = 2; value an unsigned long. example the following would set up a simple audio graph, featuring an audiodestinationnode with maxchannelcount of 2: var audioctx = new audiocontext(); var source = audioctx.createmediaelementsource(mymediaelement); source.connect(gainnode); audioctx.destination.maxchannelcount = 2; gainnode.connect(audioctx.destination); to see a more complete implementation, check out one of our mdn web audio examples, such as voice-change-o-matic or violent theremin. specification...
...ew audiocontext(); var source = audioctx.createmediaelementsource(mymediaelement); source.connect(gainnode); audioctx.destination.maxchannelcount = 2; gainnode.connect(audioctx.destination); to see a more complete implementation, check out one of our mdn web audio examples, such as voice-change-o-matic or violent theremin. specifications specification status comment web audio apithe definition of 'maxchannelcount' in that specification. working draft ...

AudioDestinationNode - Web APIs
the audiodestinationnode interface represents the end destination of an audio graph in a given context — usually the speakers of your device. it can also be the node that will "record" the audio data when used with an offlineaudiocontext. 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 number of channels in the input must be between 0 and the maxchannelcount value or an exception is raised. the audiodestinationnode of a given audiocontext can be retrieved using the audiocontext.destination property. number of inputs 1 number of outputs 0 channel count mode "explicit" channel count 2 channel interpretation "speakers" properties inherit...
...he maxchannelcount value or an exception is raised. the audiodestinationnode of a given audiocontext can be retrieved using the audiocontext.destination property. number of inputs 1 number of outputs 0 channel count mode "explicit" channel count 2 channel interpretation "speakers" properties inherits properties from its parent, audionode. audiodestinationnode.maxchannelcount is an unsigned long defining the maximum number of channels that the physical device can handle. methods no specific method; inherits methods from its parent, audionode. example there is no complex set up for using an audiodestinationnode — by default, this simply represents the output of the user's system (e.g. their speakers), so you can get ...
...onnode.maxchannelcount is an unsigned long defining the maximum number of channels that the physical device can handle. methods no specific method; inherits methods from its parent, audionode. example there is no complex set up for using an audiodestinationnode — by default, this simply represents the output of the user's system (e.g. their speakers), so you can get it hooked up inside an audio graph using only a few lines of code: var audioctx = new audiocontext(); var source = audioctx.createmediaelementsource(mymediaelement); source.connect(gainnode); gainnode.connect(audioctx.destination); to see a more complete implementation, check out one of our mdn web audio examples, such as voice-change-o-matic or violent theremin. specifications specification status comment...
...nly a few lines of code: var audioctx = new audiocontext(); var source = audioctx.createmediaelementsource(mymediaelement); source.connect(gainnode); gainnode.connect(audioctx.destination); to see a more complete implementation, check out one of our mdn web audio examples, such as voice-change-o-matic or violent theremin. specifications specification status comment web audio apithe definition of 'audiodestinationnode' in that specification. working draft ...

AudioTrackList: change event - Web APIs
the change event is fired when an audio track is enabled or disabled, for example by changing the track's enabled property. ...
... the change event is fired when an audio track is enabled or disabled, for example by changing the track's enabled property. ...
...listener(): const videoelement = document.queryselector('video'); videoelement.audiotracks.addeventlistener('change', (event) => { console.log(`'${event.type}' event fired`); }); // changing the value of `enabled` will trigger the `change` event const toggletrackbutton = document.queryselector('.toggle-track'); toggletrackbutton.addeventlistener('click', () => { const track = videoelement.audiotracks[0]; track.enabled = !track.enabled; }); using the onchange event handler property: const videoelement = document.queryselector('video'); videoelement.audiotracks.onchange = (event) => { console.log(`'${event.type}' event fired`); }; // changing the value of `enabled` will trigger the `change` event const toggletrackbutton = document.queryselector('.toggle-track'); toggletrackbutto...
... living standard ...

AudioTrackList.onaddtrack - Web APIs
the audiotracklist property onaddtrack is an event handler which is called when the addtrack event occurs, indicating that a new audio track has been added to the media element whose audio tracks the audiotracklist represents. the event is passed into the event handler in the form of a trackevent object, whose track property identifies the newly-added track. note: you can also add a handler for the addtrack event using addeventlistener(). syntax audiotracklist.onaddtrack = eventhandler; value set onaddtrack to a function that accepts as input a trackevent object which indicates in its track property which audio track has been added to the media. ...
... note: you can also add a handler for the addtrack event using addeventlistener(). syntax audiotracklist.onaddtrack = eventhandler; value set onaddtrack to a function that accepts as input a trackevent object which indicates in its track property which audio track has been added to the media. usage notes the addtrack event is called whenever a new track is added to the media element whose audio tracks are represented by the audiotracklist object. this happens when tracks are added to the element when the media is first attached to the element; one addtrack event will occur for each audio track in the media resource. ...
...w track is added to the media element whose audio tracks are represented by the audiotracklist object. this happens when tracks are added to the element when the media is first attached to the element; one addtrack event will occur for each audio track in the media resource. example this snippet establishes a handler for the addtrack event that calls a function, addtotracklist(), passing in the audiotrack object representing the newly-added track. in this scenario, that function's role is to add the new track to a list of audio tracks available to choose from. ...
...one addtrack event will occur for each audio track in the media resource. example this snippet establishes a handler for the addtrack event that calls a function, addtotracklist(), passing in the audiotrack object representing the newly-added track. in this scenario, that function's role is to add the new track to a list of audio tracks available to choose from. document.queryselector("video").audiotracks.onaddtrack = function(event) { addtotracklist(event.track); }; specifications specification status comment html living standardthe definition of 'audiotracklist.onaddtrack' in that specification. living standard html5the definition of 'audiotracklist.onaddtrack' in that specification. recommendation ...

AudioTrackList.onchange - Web APIs
the audiotracklist property onchange is an event handler which is called when the change event occurs, indicating that one or more of the audiotracks in the audiotracklist have been enabled or disabled. the event is passed into the event handler in the form of an event object; the event doesn't provide any additional information. to determine the new state of media's tracks, you'll have to look at their audiotrack.enabled flags. note: you can also add a handler for the change event using addeventlistener(). syntax audiotracklist.onchange = eventhandler; value set onchange to a function that should be called whenever tracks are enabled or disabled on the media element. example this snippet establishes a handler for the change event that looks at each of the tracks in the list, calling...
...dler; value set onchange to a function that should be called whenever tracks are enabled or disabled on the media element. example this snippet establishes a handler for the change event that looks at each of the tracks in the list, calling a function to update the state of a user interface control that indicates the current state of the track. ...
... recommendation ...
...face control using the track's id (perhaps the app uses the track id as the control element's id) and the track's enabled flag to determine which state the control should be in now. specifications specification status comment html living standardthe definition of 'audiotracklist.onchange' in that specification. living standard html5the definition of 'audiotracklist.onchange' in that specification. recommendation ...

BaseAudioContext.createAnalyser() - Web APIs
the createanalyser() method of the baseaudiocontext interface creates an analysernode, which can be used to expose audio time and frequency data and create data visualisations. note: for more on using this node, see the analysernode page. syntax var analysernode = baseaudiocontext.createanalyser(); returns an analysernode. example the following example shows basic usage of an audiocontext to create an analyser node, then use requestanimationframe() to collect time domain data repeatedly and draw an "oscilloscope style" output of the current audio input. for more complete applied examples/information, check out our voice-change-o-matic demo (see app.js lines 128–205 for relevant code). ...
...rns an analysernode. example the following example shows basic usage of an audiocontext to create an analyser node, then use requestanimationframe() to collect time domain data repeatedly and draw an "oscilloscope style" output of the current audio input. for more complete applied examples/information, check out our voice-change-o-matic demo (see app.js lines 128–205 for relevant code). var audioctx = new (window.audiocontext || window.webkitaudiocontext)(); var analyser = audioctx.createanalyser(); ... ...
... voice-change-o-matic demo (see app.js lines 128–205 for relevant code). var audioctx = new (window.audiocontext || window.webkitaudiocontext)(); var analyser = audioctx.createanalyser(); ... ...
... working draft ...

BaseAudioContext.createPeriodicWave() - Web APIs
parameters real an array of cosine terms (traditionally the a terms). imag an array of sine terms (traditionally the b terms). the real and imag arrays have to have the same length, otherwise an error is thrown. ...
... parameters real an array of cosine terms (traditionally the a terms). imag an array of sine terms (traditionally the b terms). the real and imag arrays have to have the same length, otherwise an error is thrown. ...
...: if set to true, normalization is disabled for the periodic wave. the default is false. if normalized, the resulting wave will have a maximum absolute peak value of 1. example the following example illustrates simple usage of createperiodicwave(), to create a periodicwave object containing a simple sine wave. ...
...cients of the fourier transform should be given in ascending order (i.e. (a+bi)ei,(c+di)e2i,(f+gi)e3i\left(a+bi\right)e^{i} , \left(c+di\right)e^{2i} , \left(f+gi\right)e^{3i} etc.) and can be positive or negative. a simple way of manually obtaining such coefficients (though not the best) is to use a graphing calculator. specifications specification status comment web audio apithe definition of 'createperiodicwave' in that specification. working draft ...

BaseAudioContext.destination - Web APIs
the destination property of the baseaudiocontext interface returns an audiodestinationnode representing the final destination of all audio in the context. it often represents an actual audio-rendering device such as your device's speakers. syntax baseaudiocontext.destination; value an audiodestinationnode. example note: for a full example implementation, see one of our web audio demos on the mdn github repo, like voice-change-o-matic. ...
...face returns an audiodestinationnode representing the final destination of all audio in the context. it often represents an actual audio-rendering device such as your device's speakers. syntax baseaudiocontext.destination; value an audiodestinationnode. example note: for a full example implementation, see one of our web audio demos on the mdn github repo, like voice-change-o-matic. ...
...estinationnode representing the final destination of all audio in the context. it often represents an actual audio-rendering device such as your device's speakers. syntax baseaudiocontext.destination; value an audiodestinationnode. example note: for a full example implementation, see one of our web audio demos on the mdn github repo, like voice-change-o-matic. ...
...he final destination of all audio in the context. it often represents an actual audio-rendering device such as your device's speakers. syntax baseaudiocontext.destination; value an audiodestinationnode. example note: for a full example implementation, see one of our web audio demos on the mdn github repo, like voice-change-o-matic. var audiocontext = window.audiocontext || window.webkitaudiocontext; var audioctx = new audiocontext(); // older webkit/blink browsers require a prefix var oscillatornode = audioctx.createoscillator(); var gainnode = audioctx.creategain(); oscillatornode.connect(gainnode); gainnode.connect(audioctx.destination); specifications specification status comment web audio apithe definition of 'destination' in that specification. wo...

MediaStream.getAudioTracks() - Web APIs
the getaudiotracks() method of the mediastream interface returns a sequence that represents all the mediastreamtrack objects in this stream's track set where mediastreamtrack.kind is audio. syntax var mediastreamtracks = mediastream.getaudiotracks() parameters none. return value an array of mediastreamtrack objects, one for each audio track contained in the stream. audio tracks are those tracks whose kind property is audio. the array is empty if the stream contains no audio tracks. note: the order of the returned tracks is not defined by the specification and may, in fact, change from one call to getaudiotracks() to the next. ...
...rray of mediastreamtrack objects, one for each audio track contained in the stream. audio tracks are those tracks whose kind property is audio. the array is empty if the stream contains no audio tracks. note: the order of the returned tracks is not defined by the specification and may, in fact, change from one call to getaudiotracks() to the next. early versions of this api included a special audiostreamtrack interface which was used as the type for each entry in the list of audio streams; however, this has since been merged into the main mediastreamtrack interface. example this example gets a webcam's audio and video in a stream using getusermedia(), attaches the stream to a <video> element, then sets a timer that, upon expiring, will stop the first audio track found on the stream. navi...
...sed as the type for each entry in the list of audio streams; however, this has since been merged into the main mediastreamtrack interface. example this example gets a webcam's audio and video in a stream using getusermedia(), attaches the stream to a <video> element, then sets a timer that, upon expiring, will stop the first audio track found on the stream. ...
...rack interface. example this example gets a webcam's audio and video in a stream using getusermedia(), attaches the stream to a <video> element, then sets a timer that, upon expiring, will stop the first audio track found on the stream. navigator.mediadevices.getusermedia({audio: true, video: true}) .then(mediastream => { document.queryselector('video').srcobject = mediastream; // stop the audio stream after 5 seconds settimeout(() => { const tracks = mediastream.getaudiotracks() tracks[0].stop() }, 5000) }) specifications specification status comment media capture and streamsthe definition of 'getaudiotracks()' in that specification. candidate recommendation initial definition. ...

NotifyAudioAvailableEvent - Web APIs
non-standard this feature is non-standard and is not on a standards track. do not use it on production sites facing the web: it will not work for every user. there may also be large incompatibilities between implementations and the behavior may change in the future. the non-standard, obsolete, notifyaudioavailableevent interface defines the event sent to audio elements when the audio buffer is full. ...
... facing the web: it will not work for every user. there may also be large incompatibilities between implementations and the behavior may change in the future. the non-standard, obsolete, notifyaudioavailableevent interface defines the event sent to audio elements when the audio buffer is full. properties framebuffer read only a float32array containing the raw 32-bit floating-point audio data obtained from decoding the audio (e.g., the raw data being sent to the audio hardware vs. encoded audio). ...
...ge in the future. the non-standard, obsolete, notifyaudioavailableevent interface defines the event sent to audio elements when the audio buffer is full. properties framebuffer read only a float32array containing the raw 32-bit floating-point audio data obtained from decoding the audio (e.g., the raw data being sent to the audio hardware vs. encoded audio). the data is a series of audio samples, each sample containing one 32-bit value per audio channel. all audio frames are normalized to contain 1024 samples by default, but could be any length between 512 and 16384 samples if the user has set a different length using the mozframebufferlength attribute. time a floating-point value indicating the time in seconds at which the first sample in the framebuffer occurs, relative to th...
...sample containing one 32-bit value per audio channel. all audio frames are normalized to contain 1024 samples by default, but could be any length between 512 and 16384 samples if the user has set a different length using the mozframebufferlength attribute. time a floating-point value indicating the time in seconds at which the first sample in the framebuffer occurs, relative to the start of the audio track. ...

mozbrowseraudioplaybackchange
non-standard this feature is non-standard and is not on a standards track. do not use it on production sites facing the web: it will not work for every user. there may also be large incompatibilities between implementations and the behavior may change in the future. the mozbrowseraudioplaybackchange event is fired when audio starts or stops playing within a browser <iframe>. general info specification non standard interface customevent bubbles yes cancelable yes target <iframe> default action none properties property type description target read only eventtarget the browser iframe type read only domstring the type of event. bubbles read only boolean whether the event normally bubb...
... property type description target read only eventtarget the browser iframe type read only domstring the type of event. bubbles read only boolean whether the event normally bubbles or not. cancelable read only boolean whether the event is cancellable or not. ...
... domstring the type of event. bubbles read only boolean whether the event normally bubbles or not. cancelable read only boolean whether the event is cancellable or not. details read only boolean indicates whether audio is playing in the browser. ...

AudioBuffer.copyFromChannel() - Web APIs
startinchannel optional an optional offset into the source channel's buffer from which to begin copying samples. if not specified, a value of 0 (the beginning of the buffer) is assumed by default. return value undefined. ...
... startinchannel optional an optional offset into the source channel's buffer from which to begin copying samples. if not specified, a value of 0 (the beginning of the buffer) is assumed by default. return value undefined. ...
...hat is outside the accepted range: the value of channelnumber specifies a channel number which doesn't exist (that is, it's greater than or equal to the value of numberofchannels on the channel). the value of startinchannel is outside the current range of samples that already exist in the source buffer; that is, it's greater than its current length. example this example creates a new audio buffer, then copies the samples from another channel into it. var myarraybuffer = audioctx.createbuffer(2, framecount, audioctx.samplerate); var anotherarray = new float32array(length); myarraybuffer.copyfromchannel(anotherarray, 1, 0); specification specification status comment web audio apithe definition of 'copyfromchannel' in that specification. working draf...

AudioConfiguration - Web APIs
the audioconfiguration dictionary of the media capabilities api defines the audio file being tested when calling mediacapabilities.encodinginfo() or mediacapabilities.decodinginfo() to query whether a specific audio configuration is supported, smooth, and/or power efficient. properties the audioconfiguration dictionary is made up of four audio properties, including: contenttype: a valid audio mime type, for information on possible values and what they mean, see the web audio codec guide. channels: the number of channels used by the audio track. bitrate: the number of bits used to encode one second of the audio file. samplerate: the number of audio samples making up one second of the audio file. examples //create media configuration to be tested const mediaconfig = { type : 'fi...
... mean, see the web audio codec guide. channels: the number of channels used by the audio track. bitrate: the number of bits used to encode one second of the audio file. samplerate: the number of audio samples making up one second of the audio file. examples //create media configuration to be tested const mediaconfig = { type : 'file', // 'record', 'transmission', or 'media-source' audio : { contenttype : "audio/ogg", // valid content type channels : 2, // audio channels used by the track bitrate : 132700, // number of bits used to encode 1s of audio samplerate : 5200 // number of audio samples making up that 1s. } }; // check support and performance navigator.mediacapabilities.decodinginfo(mediaconfig).then(result => { console.log('t...
... draft initial definition ...

AudioContext.close() - Web APIs
the close() method of the audiocontext interface closes the audio context, releasing any system audio resources that it uses. closed contexts cannot have new nodes created, but can decode audio data, create buffers, etc. this function does not automatically release all audiocontext-created objects, unless other references have been released as well; however, it will forcibly release any system audio resources that might prevent additional audiocontexts from being created and used, suspend the progression of audio time in the audio context, and stop processing audio data. the returned promise resolves when all audiocontext-creation-blocking resources have been released. this method throws an invalid_state_err exception if called on an offlineaudiocontext. ...
...release any system audio resources that might prevent additional audiocontexts from being created and used, suspend the progression of audio time in the audio context, and stop processing audio data. the returned promise resolves when all audiocontext-creation-blocking resources have been released. this method throws an invalid_state_err exception if called on an offlineaudiocontext. syntax var audioctx = new audiocontext(); audioctx.close().then(function() { ... }); await audioctx.close(); returns a promise that resolves with void. example the following snippet is taken from our audiocontext states demo (see it running live.) when the stop button is clicked, close() is called. when the promise resolves, the example is reset to its beginning state. ...
...audioctx = new audiocontext(); audioctx.close().then(function() { ... }); await audioctx.close(); returns a promise that resolves with void. example the following snippet is taken from our audiocontext states demo (see it running live.) when the stop button is clicked, close() is called. when the promise resolves, the example is reset to its beginning state. stopbtn.onclick = function() { audioctx.close().then(function() { startbtn.removeattribute('disabled'); susresbtn.setattribute('disabled', 'disabled'); stopbtn.setattribute('disabled', 'disabled'); }); } specifications specification status comment web audio apithe definition of 'close()' in that specification. working draft ...

AudioContext.createJavaScriptNode() - Web APIs
the audiocontext.createjavascriptnode() method creates a javascriptnode which is used for directly manipulating audio data with javascript. important: this method is obsolete, and has been renamed to audiocontext.createscriptprocessor(). see also scriptprocessornode. syntax var jsnode = audioctx.createjavascriptnode(buffersize, numinputchannels, numoutputchannels); parameters buffersize the buffer size must be in units of sample frames, i.e., one of: 256, 512, 1024, 2048, 4096, 8192, or 16384. it controls the frequency of callbacks asking for a buffer refill. smaller sizes allow for lower latency and higher for better overall quality. numinputchannels the number of input channels in the audio stream. numoutputchannels the number of output channels in the audio stream. exam...
...channels the number of input channels in the audio stream. numoutputchannels the number of output channels in the audio stream. ...
...) { that.process(e) }; } sinewave.prototype.process = function(e) { var data = e.outputbuffer.getchanneldata(0); for (var i = 0; i < data.length; ++i) { data[i] = math.sin(this.x++); } } sinewave.prototype.play = function() { this.node.connect(this.context.destination); } sinewave.prototype.pause = function() { this.node.disconnect(); } see also generating tones with the web audio api exploring the html5 web audio: visualizing sound ...

AudioContext.resume() - Web APIs
the resume() method of the audiocontext interface resumes the progression of time in an audio context that has previously been suspended. this method will cause an invalid_state_err exception to be thrown if called on an offlineaudiocontext. syntax completepromise = audiocontext.resume(); parameters none. return value a promise that resolves when the context has resumed. the promise is rejected if the context has already been closed. example the following snippet is taken from our audiocontext states demo (see it running live.) when the suspend/resume button is clicked, the audiocontext.state is queried — if it is running, suspend() is called; if it is suspended, resume() is called. in each case, the text label of the button is updated as appropriate once the promise resolves. sus...
...losed. example the following snippet is taken from our audiocontext states demo (see it running live.) when the suspend/resume button is clicked, the audiocontext.state is queried — if it is running, suspend() is called; if it is suspended, resume() is called. in each case, the text label of the button is updated as appropriate once the promise resolves. ...
...et is taken from our audiocontext states demo (see it running live.) when the suspend/resume button is clicked, the audiocontext.state is queried — if it is running, suspend() is called; if it is suspended, resume() is called. in each case, the text label of the button is updated as appropriate once the promise resolves. susresbtn.onclick = function() { if(audioctx.state === 'running') { audioctx.suspend().then(function() { susresbtn.textcontent = 'resume context'; }); } else if(audioctx.state === 'suspended') { audioctx.resume().then(function() { susresbtn.textcontent = 'suspend context'; }); } } specifications specification status comment web audio apithe definition of 'resume()' in that specification. working draft ...

AudioNode.channelCount - Web APIs
the channelcount property of the audionode interface represents an integer used to determine how many channels are used when up-mixing and down-mixing connections to any inputs to the node. channelcount's usage and precise definition depend on the value of audionode.channelcountmode: it is ignored if the channelcountmode value is max. it is used as a maximum value if the channelcountmode value is clamped-max. it is used as the exact value if the channelcountmode value is explicit. syntax var oscillator = audioctx.createoscillator(); var channels = oscillator.channelcount; value an integer. ...
...e definition depend on the value of audionode.channelcountmode: it is ignored if the channelcountmode value is max. it is used as a maximum value if the channelcountmode value is clamped-max. it is used as the exact value if the channelcountmode value is explicit. syntax var oscillator = audioctx.createoscillator(); var channels = oscillator.channelcount; value an integer. ...
... the value of audionode.channelcountmode: it is ignored if the channelcountmode value is max. it is used as a maximum value if the channelcountmode value is clamped-max. it is used as the exact value if the channelcountmode value is explicit. syntax var oscillator = audioctx.createoscillator(); var channels = oscillator.channelcount; value an integer. example var audiocontext = window.audiocontext || window.webkitaudiocontext; var audioctx = new audiocontext(); var oscillator = audioctx.createoscillator(); var gainnode = audioctx.creategain(); oscillator.connect(gainnode); gainnode.connect(audioctx.destination); oscillator.channelcount; specifications specification status comment web audio apithe definition of 'channelcount' in that specification. w...

AudioNodeOptions - Web APIs
the audionodeoptions dictionary of the web audio api specifies options that can be used when creating new audionode objects. audionodeoptions is inherited from by the option objects of the different types of audio node constructors. see for example analysernode.analysernode or gainnode.gainnode. syntax var audionodeoptions = { "channelcount" : 2, "channelcountmode" : "max", "channelinterpretation" : "discrete" } properties channelcount optional represents an integer used to determine how many channels are used when up-mixing and down-mixing connections to any inputs to the node. (see audionode.channelcount for more information.) its usage and precise definition depend on the value of audionodeoptions.channelcountmode. channelcountmode optional represents an enumerated value des...
... to determine how many channels are used when up-mixing and down-mixing connections to any inputs to the node. (see audionode.channelcount for more information.) its usage and precise definition depend on the value of audionodeoptions.channelcountmode. channelcountmode optional represents an enumerated value describing the way channels must be matched between the node's inputs and outputs. (see audionode.channelcountmode for more information including default values.) channelinterpretation optional represents an enumerated value describing the meaning of the channels. this interpretation will define how audio up-mixing and down-mixing will happen. ...
...ribing the way channels must be matched between the node's inputs and outputs. (see audionode.channelcountmode for more information including default values.) channelinterpretation optional represents an enumerated value describing the meaning of the channels. this interpretation will define how audio up-mixing and down-mixing will happen. the possible values are "speakers" or "discrete". (see audionode.channelcountmode for more information including default values.) specifications specification status comment web audio apithe definition of 'audionodeoptions' in that specification. working draft initial definition. ...

AudioScheduledSourceNode.stop() - Web APIs
the stop() method on audioscheduledsourcenode schedules a sound to cease playback at the specified time. if no time is specified, then the sound stops playing immediately. each time you call stop() on the same node, the specified time replaces any previously-scheduled stop time that hasn't occurred yet. if the node has already stopped, this method has no effect. note: if a scheduled stop time occurs before the node's scheduled start time, the node never starts to play. syntax audioscheduledsourcenode.stop([when]); parameters when optional the time, in seconds, at which the sound should stop playing. this value is specified in the same time coordinate system as the audiocontext is using for its currenttime attribute. omitting this parameter, specifying a value of 0, or passing a neg...
...tive value causes the sound to stop playback immediately. return value undefined exceptions invalidstatenode the node has not been started by calling start(). rangeerror the value specified for when is negative. example this example demonstrates starting an oscillator node, scheduled to begin playing at once and to stop playing in one second. ...
...p playback immediately. return value undefined exceptions invalidstatenode the node has not been started by calling start(). rangeerror the value specified for when is negative. example this example demonstrates starting an oscillator node, scheduled to begin playing at once and to stop playing in one second. the stop time is determined by taking the audio context's current time from audiocontext.currenttime and adding 1 second. context = new audiocontext(); osc = context.createoscillator(); osc.connect(context.destination); /* let's play a sine wave for one second. */ osc.start(); osc.stop(context.currenttime + 1); specifications specification status comment web audio apithe definition of 'stop()' in that specification. working draft ...

AudioTrackList.length - Web APIs
the read-only audiotracklist property length returns the number of entries in the audiotracklist, each of which is an audiotrack representing one audio track in the media element. a value of 0 indicates that there are no audio tracks in the media. syntax var trackcount = audiotracklist.length; value a number indicating how many audio tracks are included in the audiotracklist. each track can be accessed by treating the audiotracklist as an array of objects of type audiotrack. example this snippet gets the number of audio tracks in the first <video> element found in the dom by queryselector(). ...
...rackcount = audiotracklist.length; value a number indicating how many audio tracks are included in the audiotracklist. each track can be accessed by treating the audiotracklist as an array of objects of type audiotrack. example this snippet gets the number of audio tracks in the first <video> element found in the dom by queryselector(). var videoelem = document.queryselector("video"); var numaudiotracks = 0; if (videoelem.audiotracks) { numaudiotracks = videoelem.audiotracks.length; } note that this sample checks to be sure htmlmediaelement.audiotracks is defined, to avoid failing on browsers without support for audiotrack. specifications specification status comment html living standardthe definition of 'audiotracklist.length' in that specification. l...
...racks = videoelem.audiotracks.length; } note that this sample checks to be sure htmlmediaelement.audiotracks is defined, to avoid failing on browsers without support for audiotrack. specifications specification status comment html living standardthe definition of 'audiotracklist.length' in that specification. living standard html5the definition of 'audiotracklist.length' in that specification. recommendation ...

AudioTrackList.onremovetrack - Web APIs
the audiotracklist onremovetrack event handler is called when the removetrack event occurs, indicating that an audio track has been removed from the media element, and therefore also from the audiotracklist. the event is passed into the event handler in the form of a trackevent object, whose track property identifies the track that was removed from the media element's audiotracklist. note: you can also add a handler for the removetrack event using addeventlistener(). syntax audiotracklist.onremovetrack = eventhandler; value set onremovetrack to a function that accepts as input a trackevent object which indicates in its track property which audio track has been removed from the media element. ...
...diotracklist. note: you can also add a handler for the removetrack event using addeventlistener(). syntax audiotracklist.onremovetrack = eventhandler; value set onremovetrack to a function that accepts as input a trackevent object which indicates in its track property which audio track has been removed from the media element. example this simple example just fetches the current number of audio tracks in the media element whenever a track is removed from the media element. document.queryselector("my-video").audiotracks.onremovetrack = function(event) { mytrackcount = document.queryselector("my-video").audiotracks.length; }; the current number of audio tracks remaining in the media element is obtained from audiotracklist property length. specifications specification st...
...dia element. document.queryselector("my-video").audiotracks.onremovetrack = function(event) { mytrackcount = document.queryselector("my-video").audiotracks.length; }; the current number of audio tracks remaining in the media element is obtained from audiotracklist property length. specifications specification status comment html living standardthe definition of 'audiotracklist.onremovetrack' in that specification. living standard html5the definition of 'audiotracklist.onremovetrack' in that specification. recommendation ...

BaseAudioContext.createConstantSource() - Web APIs
working draft initial definition. browser compatibility the compatibility table on this page is generated from structured data. ...
... working draft initial definition. browser compatibility the compatibility table on this page is generated from structured data. ...
... working draft initial definition. browser compatibility the compatibility table on this page is generated from structured data. if you'd like to contribute to the data, please check out https://github.com/mdn/browser-compat-data and send us a pull request. ...

BaseAudioContext.createWaveShaper() - Web APIs
the createwaveshaper() method of the baseaudiocontext interface creates a waveshapernode, which represents a non-linear distortion. it is used to apply distortion effects to your audio. syntax baseaudioctx.createwaveshaper(); returns a waveshapernode. example the following example shows basic usage of an audiocontext to create a wave shaper node. for applied examples/information, check out our voice-change-o-matic demo (see app.js for relevant code). note: sigmoid functions are commonly used for distortion curves because of their natural properties. their s-shape, for instance, helps create a smoother sounding result. we found the below distortion curve code on stack overflow. ...
...usage of an audiocontext to create a wave shaper node. for applied examples/information, check out our voice-change-o-matic demo (see app.js for relevant code). note: sigmoid functions are commonly used for distortion curves because of their natural properties. their s-shape, for instance, helps create a smoother sounding result. we found the below distortion curve code on stack overflow. var audioctx = new (window.audiocontext || window.webkitaudiocontext)(); var distortion = audioctx.createwaveshaper(); ... function makedistortioncurve(amount) { var k = typeof amount === 'number' ? amount : 50, n_samples = 44100, curve = new float32array(n_samples), deg = math.pi / 180, i = 0, x; for ( ; i < n_samples; ++i ) { x = i * 2 / n_samples - 1; curve[i] = ( 3 +...
... working draft ...

BaseAudioContext.currentTime - Web APIs
the currenttime read-only property of the baseaudiocontext interface returns a double representing an ever-increasing hardware timestamp in seconds that can be used for scheduling audio playback, visualizing timelines, etc. it starts at 0. syntax var curtime = baseaudiocontext.currenttime; example var audiocontext = window.audiocontext || window.webkitaudiocontext; var audioctx = new audiocontext(); // older webkit/blink browsers require a prefix ... console.log(audioctx.currenttime); reduced time precision to offer protection against timing attacks and fingerprinting, the precision of audioctx.currenttime might get rounded depending on browser settings. in firefox, the privacy.reducetimerprecision preference is enabled by default and defaults to 20us in firefox 59; in 60 it will...
... in firefox, you can also enabled privacy.resistfingerprinting, the precision will be 100ms or the value of privacy.resistfingerprinting.reducetimerprecision.microseconds, whichever is larger. ...
... precision of audioctx.currenttime might get rounded depending on browser settings. in firefox, the privacy.reducetimerprecision preference is enabled by default and defaults to 20us in firefox 59; in 60 it will be 2ms. // reduced time precision (2ms) in firefox 60 audioctx.currenttime; // 23.404 // 24.192 // 25.514 // ... // reduced time precision with `privacy.resistfingerprinting` enabled audioctx.currenttime; // 49.8 // 50.6 // 51.7 // ... in firefox, you can also enabled privacy.resistfingerprinting, the precision will be 100ms or the value of privacy.resistfingerprinting.reducetimerprecision.microseconds, whichever is larger. specifications specification status comment web audio apithe definition of 'currenttime' in that specification. working draft ...

msAudioDeviceType - Web APIs
non-standard this feature is non-standard and is not on a standards track. do not use it on production sites facing the web: it will not work for every user. there may also be large incompatibilities between implementations and the behavior may change in the future. the msaudiodevicetype property of the html <audio> element, is a read/write proprietary attribute, specific to internet explorer and microsoft edge. msaudiodevicetype specifies the output device id that the audio will be sent to. syntax <audio src="sound.mp3" msaudiodevicetype="communications" /> by default, audio on your system will output to your default speakers and be considered a foreground element, meaning that the audio will play only when the element is active in the app. for real-time c...
...soft edge. msaudiodevicetype specifies the output device id that the audio will be sent to. syntax <audio src="sound.mp3" msaudiodevicetype="communications" /> by default, audio on your system will output to your default speakers and be considered a foreground element, meaning that the audio will play only when the element is active in the app. for real-time communications, you can use the msaudiodevicetype property with the value console, multimedia, or communications to specify where the current audio should output. value include a description of the property's value, including data type and what it represents. value description console specifies that the audio output will be sent to the console device. ...
...he msaudiodevicetype property with the value console, multimedia, or communications to specify where the current audio should output. value include a description of the property's value, including data type and what it represents. value description console specifies that the audio output will be sent to the console device. multimedia specifies that the audio output will be sent to the multimedia device. communications specifies that the audio output will be sent to the communications device. see also microsoft api extensions ...

HTMLMediaElement.audioTracks - Web APIs
the read-only audiotracks property on htmlmediaelement objects returns an audiotracklist object listing all of the audiotrack objects representing the media element's audio tracks. the media element may be either an <audio> element or a <video> element. the returned list is live; that is, as tracks are added to and removed from the media element, the list's contents change dynamically. once you have a reference to the list, you can monitor it for changes to detect when new audio tracks are added or existing ones removed. see event handlers in audiotracklist to learn more about watching for changes to a media element's track list. syntax var audiotracks = mediaelement.audiotracks; value a audiotracklist object representing the list of audio tracks included in the media element. ...
... see event handlers in audiotracklist to learn more about watching for changes to a media element's track list. syntax var audiotracks = mediaelement.audiotracks; value a audiotracklist object representing the list of audio tracks included in the media element. the list of tracks can be accessed using array notation, or using the object's gettrackbyid() method. each track is represented by a audiotrack object which provides information about the track. examples in this example, all of the audio tracks on a given element are muted. html the html establishes the element itself. <video id="video" src="somevideo.mp4"></video> javascript the javascript code handles muting the video element's audio tracks. ...
... a audiotrack object which provides information about the track. examples in this example, all of the audio tracks on a given element are muted. html the html establishes the element itself. <video id="video" src="somevideo.mp4"></video> javascript the javascript code handles muting the video element's audio tracks. var video = document.getelementbyid("video"); for (var i = 0; i < video.audiotracks.length; i += 1) { video.audiotracks[i].enabled = false; } specifications specification status comment html living standardthe definition of 'htmlmediaelement.audiotracks' in that specification. living standard no change from html5 html5the definition of 'htmlmediaelement.audiotracks' in that specification. recommendation initial definition. ...

MediaRecorder.audioBitsPerSecond - Web APIs
working draft initial definition. browser compatibility the compatibility table in this page is generated from structured data. if you'd like to contribute to the data, please check out https://github.com/mdn/browser-compat-data and send us a pull request. ...
... working draft initial definition. browser compatibility the compatibility table in this page is generated from structured data. if you'd like to contribute to the data, please check out https://github.com/mdn/browser-compat-data and send us a pull request. ...
...definition. browser compatibility the compatibility table in this page is generated from structured data. if you'd like to contribute to the data, please check out https://github.com/mdn/browser-compat-data and send us a pull request. ...

MediaStreamAudioDestinationNode - Web APIs
the mediastreamaudiodestinationnode interface represents an audio destination consisting of a webrtc mediastream with a single audiomediastreamtrack, which can be used in a similar way to a mediastream obtained from navigator.getusermedia. it is an audionode that acts as an audio destination, created using the audiocontext.createmediastreamdestination method. number of inputs 1 number of outputs 0 channel count 2 channel count mode "explicit" channel count interpretation "speakers" constructor mediastreamaudiodestinationnode.mediastreamaudiodestinationnode() creates a new mediastreamaudiodestinationnode object instance. properties inherits properties from its parent, audionode. ...
...er of inputs 1 number of outputs 0 channel count 2 channel count mode "explicit" channel count interpretation "speakers" constructor mediastreamaudiodestinationnode.mediastreamaudiodestinationnode() creates a new mediastreamaudiodestinationnode object instance. properties inherits properties from its parent, audionode. mediastreamaudiodestinationnode.stream is a mediastream containing a single audiomediastreamtrack with the same number of channels as the node itself. you can use this property to get a stream out of the audio graph and feed it into another construct, such as a media recorder. methods inherits methods from its parent, audionode. example specification specification status comment w...
...tinationnode.stream is a mediastream containing a single audiomediastreamtrack with the same number of channels as the node itself. you can use this property to get a stream out of the audio graph and feed it into another construct, such as a media recorder. methods inherits methods from its parent, audionode. example specification specification status comment web audio apithe definition of 'mediastreamaudiodestinationnode' in that specification. working draft ...

RTCRtpContributingSource.audioLevel - Web APIs
the read-only audiolevel property of the rtcrtpcontributingsource interface indicates the audio level contained in the last rtp packet played from the described source. audiolevel will be the level value defined in [rfc6465] if the rfc 6465 header extension is present, and otherwise null. syntax var audiolevel = rtcrtpcontributingsource.audiolevel value a double-precision floating-point number which indicates the volume level of the audio in the most recently received rtp packet from the source described by the rtcrtpcontributingsource. this value, which is in the range 0.0 to 1.0, is on a linear scale and its value is defined in dbov, or decibels (overload). this is the amplitude relative to the point at which clipping of the audio begins to occur. a value of 1.0 represents 0 dbov (maxim...
...s (overload). this is the amplitude relative to the point at which clipping of the audio begins to occur. a value of 1.0 represents 0 dbov (maximum volume), a value of 0.0 represents silence, and a value of 0.5 represents approximately 6 db spl (decibels of sound pressure level) change in the sound pressure level from 0 dbov. for both rtcrtpcontributingsource and rtcrtpsynchronizationsource, the audio level is converted to this form from the values defined in the specifications corresponding to each type of source. audiolevel may be absent from rtcrtpcontributingsource objects, which indicates that no volume level was provided by the source; however, it is required and always available on all rtcrtpsynchronizationsource objects. specifications specification status comment ...
...corresponding to each type of source. audiolevel may be absent from rtcrtpcontributingsource objects, which indicates that no volume level was provided by the source; however, it is required and always available on all rtcrtpsynchronizationsource objects. specifications specification status comment webrtc 1.0: real-time communication between browsersthe definition of 'audiolevel' in that specification. candidate recommendation initial definition. ...

Web Audio playbackRate explained - Developer guides
the playbackrate property of the <audio> and <video> elements allows us to change the speed, or rate, at which a piece of web audio or video is playing. this article explains playbackrate in detail. playbackrate basics let's starting by looking at a brief example of playbackrate usage: var myaudio = document.createelement('audio'); myaudio.setattribute('src','audiofile.mp3'); myaudio.playbackrate = 0.5; here we create an <audio> element, and set its src to a file of our choice. next we set playbackrate to 0.5, which represents half normal speed (the playbackrate is a multiplier applied to the original rate.) a complete example let's create a <video> element first, and set up video and playback rate controls in html: <video id="myvideo" controls> <source src="https://udn.reali...
...rate allows us to change it during media playback. there is also an event available called ratechange, which fires every time the playbackrate changes. browser support chrome 20+ ✔ firefox 20+ ✔ ie 9+ ✔ safari 6+ ✔ opera 15+ ✔ mobile chrome (android) ✖ mobile firefox 24+ ✔ ie mobile ✖ mobile safari 6+ (ios) ✔ opera mobile ✖ notes most browsers stop playing audio outside playbackrate bounds of 0.5 and 4, leaving the video playing silently. for most applications, it's recommended that you limit the range to between 0.5 and 4. the pitch of the audio track does not change when playbackrate is altered. negative values will not cause the media to play in reverse. ie9+ switches to the default playback rate when an ended event is fired. firefox generates a...
...tch of the audio track does not change when playbackrate is altered. negative values will not cause the media to play in reverse. ie9+ switches to the default playback rate when an ended event is fired. firefox generates a ratechange event when the media source is substituted. on ios 7 you can only affect the playbackrate when the media is paused (not while it's playing). ...

Media type and format guide: image, audio, and video content - Web media technologies
image file types for web designers fundamental information about the various image file types that may be useful for web designers, including best practices and use cases for each type, and guidelines for choosing the right image file format for specific types of content. media file types and codecs media containers (file types) a guide to the file types that contain media data. some are audio-specific, while others may be used for either audio or combined audiovisual content such as movies. includes overviews of each of the file types supported by the major web browsers, along with browser support information and supported features. web audio codec guide a guide to the audio codecs allowed for by the common media containers, as well as by the major browsers. includes benefits, li...
...cifying the mime type describing a media format, you can provide details using the codecs parameter as part of the type string. this guide describes the format and possible values of the codecs parameter for the common media types. codecs used by webrtc webrtc doesn't use a container, but instead streams the encoded media itself from peer to peer using mediastreamtrack objects to represent each audio or video track. this guide discusses the codecs commonly used with webrtc. guides concepts digital audio concepts an introduction to how audio is converted into digital form and stored for use by computers. it explains basic concepts about how audio is sampled, as well as concepts such as sample rate, audio frames, and audio compression. digital video concepts a guide to fundamental co...
...explains basic concepts about how audio is sampled, as well as concepts such as sample rate, audio frames, and audio compression. digital video concepts a guide to fundamental concepts involved with digital video as used on the web, including basics about color formats, chroma subsampling, how human perception influences video coding, and so forth. tutorials and how-tos learning: video and audio content this tutorial introduces and details the use of media on the web. handling media support issues in web content in this guide, we look at how to build web content that maximizes quality or performance while providing the broadest possible compatibility, by choosing media formats wisely, and offering fallbacks and alternate formats where it would be helpful. other topics media capab...

Media (Audio-visual presentation) - MDN Web Docs Glossary: Definitions of Web-related terms
the term media (more accurately, multimedia) refers to audio, video, or combined audio-visual material such as music, recorded speech, movies, tv shows, or any other form of content that is presented over a period of time. more broadly, media may include still images such as photographs or other still images. media content can be recorded, played back, presented, and at times interacted with in various ways. ...
...oadly, media may include still images such as photographs or other still images. ...

AudioContext.baseLatency - Web APIs
the baselatency read-only property of the audiocontext interface returns a double that represents the number of seconds of processing latency incurred by the audiocontext passing an audio buffer from the audiodestinationnode — i.e. the end of the audio graph — into the host system's audio subsystem ready for playing. note: you can request a certain latency during construction time with the latencyhint option, but the browser may ignore the option. syntax var baselatency = audioctx.baselatency; value a double representing the base latency in seconds. ...
...onnode — i.e. the end of the audio graph — into the host system's audio subsystem ready for playing. note: you can request a certain latency during construction time with the latencyhint option, but the browser may ignore the option. syntax var baselatency = audioctx.baselatency; value a double representing the base latency in seconds. example // default latency ("interactive") const audioctx1 = new audiocontext(); console.log(audioctx1.baselatency); // 0.00 // higher latency ("playback") const audioctx2 = new audiocontext({ latencyhint: 'playback' }); console.log(audioctx2.baselatency); // 0.15 specifications specification status comment web audio apithe definition of 'baselatency' in that specification. working draft initial definition. br...

AudioContextOptions.latencyHint - Web APIs
the audiocontextoptions dictionary (used when instantiating an audiocontext) may contain a property named latencyhint, which indicates the preferred maximum latency in seconds for the audio context. the value is specified either as a member of the string enum audiocontextlatencycategory or a double-precision value. syntax audiocontextoptions.latencyhint = "interactive"; audiocontextoptions.latencyhint = 0.2; var latencyhint = audiocontextoptions.latencyhint; value the preferred maximum latency for the audiocontext. there are two ways this value can be specified. the best way to specify the preferred latency is to use a value form the string enum audiocontextlatencycategory. in fact, the default value of latencyhint is "interactive" (meaning the browser should try to use the lowest possib...
...use a value form the string enum audiocontextlatencycategory. in fact, the default value of latencyhint is "interactive" (meaning the browser should try to use the lowest possible and reliable latency it can). the value can also be specified as a double-precision floating-point value, specifying the preferred maximum latency in seconds. this provides more precise control over the balance between audio latency and device energy usage. to determine the actual latency of the context after creating it, check the value of the context's baselatency property. specifications specification status comment web audio apithe definition of 'audiocontextoptions.latencyhint' in that specification. working draft initial definition. ...

AudioContextOptions.sampleRate - Web APIs
the audiocontextoptions dictionary (used when instantiating an audiocontext) may contain a property named samplerate, which indicates the sample rate to use for the new context. the value must be a floating-point value indicating the sample rate, in samples per second, for which to configure the new context; additionally, the value must be one which is supported by audiobuffer.samplerate. syntax audiocontextoptions.samplerate = 44100; var samplerate = audiocontextoptions.samplerate; value the desired sample rate for the audiocontext, specified in samples per second. the value must be compatible with audiobuffer.samplerate. this value should typically be between 8,000 hz and 96,000 hz; the default will vary depending on the output device, but the sample rate 44,100 hz is the most common. ...
...ple rate for the audiocontext, specified in samples per second. the value must be compatible with audiobuffer.samplerate. this value should typically be between 8,000 hz and 96,000 hz; the default will vary depending on the output device, but the sample rate 44,100 hz is the most common. if the samplerate property is not included in the options, or the options are not specified when creating the audio context, the new context's output device's preferred sample rate is used by default. specifications specification status comment web audio apithe definition of 'audiocontextoptions.samplerate' in that specification. working draft initial definition. ...

AudioParam.cancelScheduledValues() - Web APIs
the cancelscheduledvalues() method of the audioparam interface cancels all scheduled future changes to the audioparam. syntax var audioparam = audioparam.cancelscheduledvalues(starttime) parameters starttime a double representing the time (in seconds) after the audiocontext was first created after which all scheduled changes will be cancelled. returns a reference to this audioparam object. in some older implementations this method returns void. ...
...uture changes to the audioparam. syntax var audioparam = audioparam.cancelscheduledvalues(starttime) parameters starttime a double representing the time (in seconds) after the audiocontext was first created after which all scheduled changes will be cancelled. returns a reference to this audioparam object. in some older implementations this method returns void. examples var gainnode = audioctx.creategain(); gainnode.gain.setvaluecurveattime(wavearray, audioctx.currenttime, 2); //'gain' is the audioparam gainnode.gain.cancelscheduledvalues(audioctx.currenttime); specifications specification status comment web audio apithe definition of 'cancelscheduledvalues' in that specification. working draft ...

AudioScheduledSourceNode: ended event - Web APIs
the ended event of the audioscheduledsourcenode interface is fired when the source node has stopped playing. bubbles no cancelable no interface event event handler property audioscheduledsourcenode.onended usage notes this event occurs when a audioscheduledsourcenode has stopped playing, either because it's reached a predetermined stop time, the full duration of the audio has been performed, or because the entire buffer has been played. examples in this simple example, an event listener for the ended event is set up to enable a "start" button in the user interface when the node stops playing: node.addeventlistener('ended', () => { document.getelementbyid("startbutton").disabled = false; }) you can also set up the event handler usin...
...duration of the audio has been performed, or because the entire buffer has been played. examples in this simple example, an event listener for the ended event is set up to enable a "start" button in the user interface when the node stops playing: node.addeventlistener('ended', () => { document.getelementbyid("startbutton").disabled = false; }) you can also set up the event handler using the audioscheduledsourcenode.onended property: node.onended = function() { document.getelementbyid("startbutton").disabled = false; } for an example of the ended event in use, see our audio-buffer example on github. specifications specification status comment web audio apithe definition of 'onended' in that specification. working draft initial specification. ...

AudioScheduledSourceNode.onended - Web APIs
the onended event handler for the audioscheduledsourcenode interface specifies an eventhandler to be executed when the ended event occurs on the node. this event is sent to the node when the concrete interface (such as audiobuffersourcenode, oscillatornode, or constantsourcenode) determines that it has stopped playing. the ended event is only sent to a node configured to loop automatically when the node is stopped using its stop() method. this is the case, for example, when using an audiobuffersourcenode with its loop property set to true. syntax audioscheduledsourcenode.onended = eventhandler; value a function which is called by the browser when the ended event occurs on the audioscheduledsourcenode. the function receives as input a single parameter, which is an object of type event d...
...ch is an object of type event describing the event that occurred. examples in this simple example, an event listener for the ended event is set up to enable a "start" button in the user interface when the node stops playing. node.onended = function(event) { document.getelementbyid("startbutton").disabled = false; } specifications specification status comment web audio apithe definition of 'onended' in that specification. working draft initial specification. ...

AudioWorklet - Web APIs
the audioworklet interface of the web audio api is used to supply custom audio processing scripts that execute in a separate thread to provide very low latency audio processing. the worklet's code is run in the audioworkletglobalscope global execution context, using a separate web audio thread which is shared by the worklet and other audio nodes. access the audio context's instance of audioworklet through the baseaudiocontext.audioworklet property. properties the audioworklet interface does not define any properties of its own, but does inherit properties of worklet. methods this interface inherits methods from worklet. the audioworklet interface does not define any methods of its own. events audioworklet has no events to which it responds. ...
...t's instance of audioworklet through the baseaudiocontext.audioworklet property. properties the audioworklet interface does not define any properties of its own, but does inherit properties of worklet. methods this interface inherits methods from worklet. the audioworklet interface does not define any methods of its own. events audioworklet has no events to which it responds. examples see audioworkletnode for complete examples of custom audio node creation. specifications specification status comment web audio apithe definition of 'audioworklet' in that specification. working draft initial definition. ...

BaseAudioContext.createIIRFilter() - Web APIs
the maximum length of this array is 20, and at least one value must be nonzero. feedback an array of floating-point values specifying the feedback (denominator) coefficients for the transfer function of the iir filter. this array may have up to 20 members, the first of which must not be zero. ...
...t be zero. return value an iirfilternode implementing the filter with the specified feedback and feedforward coefficient arrays. exceptions invalidstateerror all of the feedforward coefficients are 0, and/or the first feedback coefficient is 0. notsupportederror one or both of the input arrays exceeds 20 members. specifications specification status comment web audio apithe definition of 'createiirfilter()' in that specification. working draft ...

BaseAudioContext.sampleRate - Web APIs
the samplerate property of the baseaudiocontext interface returns a floating point number representing the sample rate, in samples per second, used by all nodes in this audio context. this limitation means that sample-rate converters are not supported. syntax baseaudiocontext.samplerate; value a floating point number indicating the audio context's sample rate, in samples per second. example note: for a full web audio example implementation, see one of our web audio demos on the mdn github repo, like panner-node. try entering audioctx.samplerate into your browser console. var audiocontext = window.audiocontext || window.webkitaudiocontext; var audioctx = new audiocontext(); // older webkit/blink browsers require a prefix ... ...
...ple rate, in samples per second. example note: for a full web audio example implementation, see one of our web audio demos on the mdn github repo, like panner-node. try entering audioctx.samplerate into your browser console. var audiocontext = window.audiocontext || window.webkitaudiocontext; var audioctx = new audiocontext(); // older webkit/blink browsers require a prefix ... console.log(audioctx.samplerate); specifications specification status comment web audio apithe definition of 'samplerate' in that specification. working draft ...

BaseAudioContext.state - Web APIs
the state read-only property of the baseaudiocontext interface returns the current state of the audiocontext. syntax baseaudiocontext.state; value a domstring. possible values are: suspended: the audio context has been suspended (with the audiocontext.suspend() method.) running: the audio context is running normally. closed: the audio context has been closed (with the audiocontext.close() method.) example the following snippet is taken from our audiocontext states demo (see it running live.) the audiocontext.onstatechange hander is used to log the current state to the console every time it changes. ...
...s been suspended (with the audiocontext.suspend() method.) running: the audio context is running normally. closed: the audio context has been closed (with the audiocontext.close() method.) example the following snippet is taken from our audiocontext states demo (see it running live.) the audiocontext.onstatechange hander is used to log the current state to the console every time it changes. audioctx.onstatechange = function() { console.log(audioctx.state); } specifications specification status comment web audio apithe definition of 'state' in that specification. working draft ...

MediaElementAudioSourceNode() - Web APIs
the mediaelementaudiosourcenode() constructor creates a new mediaelementaudiosourcenode object instance. syntax var myaudiosource = new mediaelementaudiosourcenode(context, options); parameters inherits parameters from the audionodeoptions dictionary. context an audiocontext representing the audio context you want the node to be associated with. options a mediaelementaudiosourceoptions dictionary object defining the properties you want the mediaelementaudiosourcenode to have: mediaelement: an htmlmediaelement that will be used as the source for the audio. return value a new mediaelementaudiosourcenode object instance. ...
... audiocontext representing the audio context you want the node to be associated with. options a mediaelementaudiosourceoptions dictionary object defining the properties you want the mediaelementaudiosourcenode to have: mediaelement: an htmlmediaelement that will be used as the source for the audio. return value a new mediaelementaudiosourcenode object instance. example var ac = new audiocontext(); var mediaelement = document.createelement('audio'); var options = { mediaelement : mediaelement } var myaudiosource = new mediaelementaudiosourcenode(ac, options); specifications specification status comment web audio apithe definition of 'mediaelementaudiosourcenode' in that specification. working draft ...

MediaElementAudioSourceNode.mediaElement - Web APIs
the mediaelementaudiosourcenode interface's read-only mediaelement property indicates the htmlmediaelement that contains the audio track from which the node is receiving audio. this stream was specified when the node was first created, either using the mediaelementaudiosourcenode() constructor or the audiocontext.createmediaelementsource() method. syntax audiosourceelement = mediaelementaudiosourcenode.mediaelement; value an htmlmediaelement representing the element which contains the source of audio for the node. ...
... which the node is receiving audio. this stream was specified when the node was first created, either using the mediaelementaudiosourcenode() constructor or the audiocontext.createmediaelementsource() method. syntax audiosourceelement = mediaelementaudiosourcenode.mediaelement; value an htmlmediaelement representing the element which contains the source of audio for the node. examples const audioctx = new window.audiocontext(); const audioelem = document.queryselector('audio'); let options = { mediaelement: audioelem } let source = new mediaelementaudiosourcenode(audioctx, options); console.log(source.mediaelement); specifications specification status comment web audio apithe definition of 'mediaelementaudiosourcenode.mediaelement' in that specification. w...

OfflineAudioCompletionEvent.renderedBuffer - Web APIs
working draft initial definition. browser compatibility the compatibility table on this page is generated from structured data. if you'd like to contribute to the data, please check out https://github.com/mdn/browser-compat-data and send us a pull request. ...
... working draft initial definition. browser compatibility the compatibility table on this page is generated from structured data. if you'd like to contribute to the data, please check out https://github.com/mdn/browser-compat-data and send us a pull request. ...

OfflineAudioCompletionEvent - Web APIs
the web audio api offlineaudiocompletionevent interface represents events that occur when the processing of an offlineaudiocontext is terminated. the complete event implements this interface. note: this interface is marked as deprecated; it is still supported for legacy reasons, but it will soon be superseded when the promise version of offlineaudiocontext.startrendering is supported in browsers, which will no longer need it. constructor offlineaudiocompletionevent.offlineaudiocompletionevent creates a new offlineaudiocompletionevent object instance. properties also inherits properties from its parent, event. offlineaudiocompletionevent.renderedbuffer read only an audiobuffer containing the result of processing an offlineaudiocontext. methods inherits methods from its parent...
...iocompletionevent creates a new offlineaudiocompletionevent object instance. properties also inherits properties from its parent, event. offlineaudiocompletionevent.renderedbuffer read only an audiobuffer containing the result of processing an offlineaudiocontext. methods inherits methods from its parent, event. specifications specification status comment web audio apithe definition of 'offlineaudiocompletionevent' in that specification. working draft ...

OfflineAudioContext.suspend() - Web APIs
the suspend() method of the offlineaudiocontext interface schedules a suspension of the time progression in the audio context at the specified time and returns a promise. this is generally useful at the time of manipulating the audio graph synchronously on offlineaudiocontext. note that the maximum precision of suspension is the size of the render quantum and the specified suspension time will be rounded down to the nearest render quantum boundary. for this reason, it is not allowed to schedule multiple suspends at the same quantized frame. also scheduling should be done while the context is not running to ensure the precise suspension. syntax offlineaudiocontext.suspend(suspendtime).then(function() { ... }); parameters suspendtime a double specifying the suspend time, in seconds. re...
...to void. exceptions the promise is rejected when any exception is encountered. invalidstateerror if the quantized frame number is one of the following: a negative number is less than or equal to the current time is greater than or equal to the total render duration is scheduled by another suspend for the same time specifications specification status comment web audio apithe definition of 'suspend()' in that specification. working draft initial definition. ...

Guide to streaming audio and video - Web media technologies
<<<...xxxxxx...>>> protocols in addition to the configuration of the server and the streaming code, there are sometimes special protocols which can be used to optimize performance. https live streaming https live streaming (hls) is a protocol developed by apple and supported by safari on all of their platforms. hls may also be supported in other environments, although in some cases this support is conditional. ...
...irefox for android does as well, in order to avoid strange compatibility errors from occurring due to this assumption being incorrect. the desktop version of android, however, does not support hls. hls uses playlists to allow the user to not only select the media to stream, but also to choose among versions or forms of the same media. for example, hls lets the server stream a video with multiple audio streams which the user can choose from, in order to hear their own language. additionally, forms of the stream can be provided which are optimized for different network conditions. in this way, live streams can be made flexible and highly performant. as of mid-2017, hls has been standardized as rfc 8217. ...

MozAudioAvailable - Archive of obsolete content
the mozaudioavailable event is fired when the audio buffer is full and the corresponding raw samples are available. general info specification dom l3 interface event bubbles no cancelable no target element default action none. properties property type description target read only eventtarget the event target (the topmost target in the dom tree). type read only domstring the type of event. bubbles read only boolean does the event normally bubble? cancelable read only boolean is it possible to cancel the event? framebuffer read only framebuffer (array) the decoded audio sample data (i.e., floats). time read only float timestamp for these samples measured fro...

AudioContext.createWaveTable() - Web APIs
it was the older method for creating a periodic waveform. ...

AudioContext.outputLatency - Web APIs
the outputlatency read-only property of the audiocontext interface provides an estimation of the output latency of the current audio context. this is the time, in seconds, between the browser passing an audio buffer out of an audio graph over to the host system's audio subsystem to play, and the time at which the first sample in the buffer is actually processed by the audio output device. it varies depending on the platform and the available hardware. syntax var outputlatency = audioctx.outputlatency; value a double representing the output latency in seconds. example const audioctx = new audiocontext(); console.log(audioctx.outputlatency); specifications specification status comment web audio apithe definition of 'outputlatency' in that specification. work...

AudioNode.numberOfInputs - Web APIs
the numberofinputs property of the audionode interface returns the number of inputs feeding the node. source nodes are defined as nodes having a numberofinputs property with a value of 0. syntax var numinputs = audionode.numberofinputs; value an integer ≥ 0. example const audioctx = new audiocontext(); const oscillator = audioctx.createoscillator(); const gainnode = audioctx.creategain(); oscillator.connect(gainnode).connect(audioctx.destination); console.log(oscillator.numberofinputs); // 0 console.log(gainnode.numberofinputs); // 1 console.log(audioctx.destination.numberofinputs); // 1 specifications specification status comment web audio apithe definition of 'numberofinputs' in that specification. working draft ...

AudioNode.numberOfOutputs - Web APIs
the numberofoutputs property of the audionode interface returns the number of outputs coming out of the node. destination nodes — like audiodestinationnode — have a value of 0 for this attribute. syntax var numoutputs = audionode.numberofoutputs; value an integer ≥ 0. example const audioctx = new audiocontext(); const oscillator = audioctx.createoscillator(); const gainnode = audioctx.creategain(); oscillator.connect(gainnode).connect(audioctx.destination); console.log(oscillator.numberofoutputs); // 1 console.log(gainnode.numberofoutputs); // 1 console.log(audioctx.destination.numberofoutputs); // 0 specifications specification status comment web audio apithe definition of 'numberofoutputs' in that specification. working draft ...

AudioParam.defaultValue - Web APIs
the defaultvalue read-only property of the audioparam interface represents the initial value of the attributes as defined by the specific audionode creating the audioparam. syntax var defaultval = audioparam.defaultvalue; value a floating-point number. example const audioctx = new audiocontext(); const gainnode = audioctx.creategain(); const defaultval = gainnode.gain.defaultvalue; console.log(defaultval); // 1 console.log(defaultval === gainnode.gain.value); // true specifications specification status comment web audio apithe definition of 'defaultvalue' in that specification. working draft ...

AudioParam.maxValue - Web APIs
the maxvalue read-only property of the audioparam interface represents the maximum possible value for the parameter's nominal (effective) range. syntax var maxval = audioparam.maxvalue; value a floating-point number indicating the maximum value permitted for the parameter's nominal range. the default value of maxvalue is the maximum positive single-precision floating-point value (+340,282,346,638,528,859,811,704,183,484,516,925,440). example const audioctx = new audiocontext(); const gainnode = audioctx.creategain(); console.log(gainnode.gain.maxvalue); // 3.4028234663852886e38 specifications specification status comment web audio apithe definition of 'maxvalue' in that specification. working draft ...

AudioParam.minValue - Web APIs
the minvalue read-only property of the audioparam interface represents the minimum possible value for the parameter's nominal (effective) range. syntax var minval = audioparam.minvalue; value a floating-point number indicating the minimum value permitted for the parameter's nominal range. the default value of minvalue is the minimum negative single-precision floating-point value (-340,282,346,638,528,859,811,704,183,484,516,925,440). example const audioctx = new audiocontext(); const gainnode = audioctx.creategain(); console.log(gainnode.gain.minvalue); // -3.4028234663852886e38 specifications specification status comment web audio apithe definition of 'minvalue' in that specification. working draft ...

AudioParamMap - Web APIs
the web audio api interface audioparammap represents a set of multiple audio parameters, each described as a mapping of a domstring identifying the parameter to the audioparam object representing its value. properties the audioparammap object is accessed as a map in which each parameter is identified by a name string which is mapped to an audioparam containing the value of that parameter. in addition, there are the following properties available: size ? methods entries() ? foreach() ? has() ? keys() ? values() ? ...

AudioTrack.id - Web APIs
the id property contains a string which uniquely identifies the track represented by the audiotrack. this id can be used with the audiotracklist.gettrackbyid() method to locate a specific track within the media associated with a media element. the track id can also be used as the fragment of a url that loads the specific track (if the media supports media fragments). syntax var trackid = audiotrack.id; value a domstring which identifies the track, suitable for use when calling gettrackbyid() on an audiotracklist such as the one specified by a media element's audiotracks property. specifications specification status comment html living standardthe definition of 'audiotrack.id' in that specification. living standard html5the definition of 'au...

AudioTrack.sourceBuffer - Web APIs
the read-only audiotrack property sourcebuffer returns the sourcebuffer that created the track, or null if the track was not created by a sourcebuffer or the sourcebuffer has been removed from the mediasource.sourcebuffers attribute of its parent media source. syntax var sourcebuffer = audiotrack.sourcebuffer; value a sourcebuffer or null. specifications specification status comment media source extensionsthe definition of 'audiotrack: sourcebuffer' in that specification. recommendation ...

AudioTrackList: addtrack event - Web APIs
the addtrack event is fired when a track is added to an audiotracklist. bubbles no cancelable no interface trackevent event handler property onaddtrack examples using addeventlistener(): const videoelement = document.queryselector('video'); videoelement.audiotracks.addeventlistener('addtrack', (event) => { console.log(`audio track: ${event.track.label} added`); }); using the onaddtrack event handler property: const videoelement = document.queryselector('video'); videoelement.audiotracks.onaddtrack = (event) => { console.log(`audio track: ${event.track.label} added`); }; specifications specification status html living standardthe definition of 'addtrack' in that specification. living standard ...

AudioTrackList: removetrack event - Web APIs
the removetrack event is fired when a track is removed from an audiotracklist. bubbles no cancelable no interface trackevent event handler property onremovetrack examples using addeventlistener(): const videoelement = document.queryselector('video'); videoelement.audiotracks.addeventlistener('removetrack', (event) => { console.log(`audio track: ${event.track.label} removed`); }); using the onremovetrack event handler property: const videoelement = document.queryselector('video'); videoelement.audiotracks.onremovetrack = (event) => { console.log(`audio track: ${event.track.label} removed`); }; specifications specification status html living standardthe definition of 'removetrack' in that specification. l...

AudioWorkletNode.onprocessorerror - Web APIs
the onprocessorerror property of the audioworkletnode interface defines an event handler function to be called when the processorerror event fires. this occurs when the underlying audioworkletprocessor behind the node throws an exception in its constructor, the process method, or any user-defined class method. once an exception is thrown, the processor (and thus the node) will output silence throughout its lifetime. syntax audioworkletnode.onprocessorerror = function() { ... }; examples // fill in example snippet specifications specification status comment web audio apithe definition of 'onprocessorerror' in that specification. working draft ...

BaseAudioContext.listener - Web APIs
the listener property of the baseaudiocontext interface returns an audiolistener object that can then be used for implementing 3d audio spatialization. syntax baseaudiocontext.listener; value an audiolistener object. example note: for a full web audio spatialization example, see our panner-node demo. var audiocontext = window.audiocontext || window.webkitaudiocontext; var audioctx = new audiocontext(); // older webkit/blink browsers require a prefix ... var mylistener = audioctx.listener; specifications specification status comment web audio apithe definition of 'listener' in that specification. working draft ...

BaseAudioContext.onstatechange - Web APIs
the onstatechange property of the baseaudiocontext interface defines an event handler function to be called when the statechange event fires: this occurs when the audio context's state changes. syntax baseaudiocontext.onstatechange = function() { ... }; example the following snippet is taken from our audiocontext states demo (see it running live.) the onstatechange hander is used to log the current state to the console every time it changes. audioctx.onstatechange = function() { console.log(audioctx.state); } specifications specification status comment web audio apithe definition of 'onstatechange' in that specification. working draft ...

HTMLMediaElement.msInsertAudioEffect() - Web APIs
non-standard this feature is non-standard and is not on a standards track. do not use it on production sites facing the web: it will not work for every user. there may also be large incompatibilities between implementations and the behavior may change in the future. the htmlmediaelement.msinsertaudioeffect() method inserts the specified audio effect into the media pipeline. this proprietary method is specific to internet explorer and microsoft edge. syntax htmlmediaelement.msinsertaudioeffect(activatableclassid: domstring, effectrequired: boolean, config); parameters activatableclassid a domstring defining the audio effects class. effectrequired a boolean which if set to true requires an audio effect to be defined. configoptional an optional object to...

audioCapabilities - Web APIs
the mediakeysystemconfiguration.audiocapabilities read-only property returns an array of supported audio type and capability pairs. syntax var audiocapabilities[ {contenttype: 'contenttype', robustness:'robustness'}] = mediasystemconfiguration.audiocapabilities; specifications specification status comment encrypted media extensionsthe definition of 'audiocapabilities' in that specification. recommendation initial definition ...

MediaStreamAudioDestinationNode.stream - Web APIs
the stream property of the audiocontext interface represents a mediastream containing a single audiomediastreamtrack with the same number of channels as the node itself. you can use this property to get a stream out of the audio graph and feed it into another construct, such as a media recorder. syntax var audioctx = new audiocontext(); var destination = audioctx.createmediastreamdestination(); var mystream = destination.stream; value a mediastream. example specifications specification status comment web audio apithe definition of 'stream' in that specification. working draft ...

MediaStreamAudioSourceOptions.mediaStream - Web APIs
the mediastreamaudiosourceoptions dictionary's mediastream property must specify the mediastream from which to retrieve audio data when instantiating a mediastreamaudiosourcenode using the mediastreamaudiosourcenode() constructor. syntax mediastreamaudiosourceoptions = { mediastream: audiosourcestream; } mediastreamaudiosourceoptions.mediastream = audiosourcestream; value a mediastream representing the stream from which to use a mediastreamtrack as the source of audio for the node. example specifications specification status comment web audio apithe definition of 'mediastreamaudiosourceoptions.mediastream' in that specification. working draft ...

MediaStreamAudioSourceOptions - Web APIs
the mediastreamaudiosourceoptions dictionary provides configuration options used when creating a mediastreamaudiosourcenode using its constructor. it is not needed when using the audiocontext.createmediastreamsource() method. properties mediastream a required property which specifies the mediastream from which to obtain audio for the node. specifications specification status comment web audio apithe definition of 'mediastreamaudiosourceoptions' in that specification. working draft ...

MediaStreamTrackAudioSourceOptions - Web APIs
the mediastreamtrackaudiosourceoptions dictionary is used when specifying options to the mediastreamtrackaudiosourcenode() constructor. it isn't needed when using the audiocontext.createmediastreamtracksource() method. properties mediastreamtrack the mediastreamtrack from which to take audio data for this node's output. specifications specification status comment web audio apithe definition of 'mediastreamtrackaudiosourceoptions' in that specification. working draft ...

OfflineAudioContext.length - Web APIs
the length property of the offlineaudiocontext interface returns an integer representing the size of the buffer in sample-frames. syntax var length = offlineaudiocontext.length; value an integer representing the size of the buffer in sample-frames. specifications specification status comment web audio apithe definition of 'length' in that specification. working draft initial definition. ...

OfflineAudioContext.oncomplete - Web APIs
the oncomplete event handler of the offlineaudiocontext interface is called when the audio processing is terminated, that is when the complete event (of type offlineaudiocompletionevent) is raised. syntax var offlineaudioctx = new offlineaudiocontext(); offlineaudioctx.oncomplete = function() { ... } example when processing is complete, you might want to use the oncomplete handler the prompt the user that the audio can now be played, and enable the play button. offlineaudioctx.oncomplete = function() { console.log('offline audio processing now complete'); showmodaldialog('song processed and ready to play'); playbtn.disabled = false; } specifications specification status comment web audio apithe definition of 'oncomplete' in that specification. working ...

OfflineAudioContext.resume() - Web APIs
the resume() method of the offlineaudiocontext interface resumes the progression of time in an audio context that has been suspended. the promise resolves immediately because the offlineaudiocontext does not require the audio hardware. if the context is not currently suspended or the rendering has not started, the promise is rejected with invalidstateerror. syntax offlineaudiocontext.resume().then(function() { ... }); parameters none. returns a promise resolving to void. exceptions the promise is rejected when the following exception is encountered. invalidstateerror if the context is not currently suspended or the rendering has not started. specifications specification status comment web audio apithe definition of 'resume()' in that specification. wo...

SourceBuffer.audioTracks - Web APIs
the audiotracks read-only property of the sourcebuffer interface returns a list of the audio tracks currently contained inside the sourcebuffer. syntax var myaudiotracks = sourcebuffer.audiotracks; value an audiotracklist object. example tbd specifications specification status comment media source extensionsthe definition of 'audiotracks' in that specification. recommendation initial definition. ...

SpeechRecognition: audioend event - Web APIs
the audioend event of the web speech api is fired when the user agent has finished capturing audio for speech recognition. bubbles no cancelable no interface event event handler onaudioend examples you can use the audioend event in an addeventlistener method: var recognition = new webkitspeechrecognition() || new speechrecognition(); recognition.addeventlistener('audioend', function() { console.log('audio capturing ended'); }); or use the onaudioend event handler property: recognition.onaudioend = function() { console.log('audio capturing ended'); } specifications specification status comment web speech apithe definition of 'speech recognition events' in that specification. draft ...

SpeechRecognition: audiostart event - Web APIs
the audiostart event of the web speech api is fired when the user agent has started to capture audio for speech recognition. bubbles no cancelable no interface event event handler onaudiostart examples you can use the audiostart event in an onaudiostart method: var recognition = new webkitspeechrecognition() || new speechrecognition(); recognition.addeventlistener('audiostart', function() { console.log('audio capturing started'); }); or use the onaudiostart event handler property: recognition.onaudiostart = function() { console.log('audio capturing started'); } specifications specification status comment web speech apithe definition of 'speech recognition events' in that specification. dra...

SpeechRecognition.onaudioend - Web APIs
the onaudioend property of the speechrecognition interface represents an event handler that will run when the user agent has finished capturing audio (when the audioend event fires.) syntax myspeechrecognition.onaudioend = function() { ... }; examples var recognition = new speechrecognition(); recognition.onaudioend = function() { console.log('audio capturing ended'); } specifications specification status comment web speech apithe definition of 'onaudioend' in that specification. draft ...

SpeechRecognition.onaudiostart - Web APIs
the onaudiostart property of the speechrecognition interface represents an event handler that will run when the user agent has started to capture audio (when the audiostart event fires.) syntax myspeechrecognition.onaudiostart = function() { ... }; examples var recognition = new speechrecognition(); recognition.onaudiostart = function() { console.log('audio capturing started'); } specifications specification status comment web speech apithe definition of 'onaudiostart' in that specification. draft ...

Tools for analyzing Web Audio usage - Web APIs
while working on your web audio api code, you may find that you need tools to analyze the graph of nodes you create or to otherwise debug your work. this article discusses tools available to help you do that. chrome a handy web audio inspector can be found in the chrome web store. edge add information for developers using microsoft edge. firefox firefox offers a native web audio editor. safari add information for developers working in safari. ...

Using audio and video in HTML - Web media technologies
the html <audio> and <video> elements let you embed audio and video content into a web page. more than that, their underlying interfaces are the key to manipulating and playing back media in general, even offscreen. in this guide, we'll review the media elements and how to use them in html content. note: this guide is a planned update to integrate content from various scattered places on mdn into one cohesive document or document set. the work is not yet scheduled but hopefully will come soon. for now, some of the key information you may be looking for can be found in our learning area article on the topic. we don't have a particularly good guide to using these objects offscreen at this time, although audio and video manipulation may be a good start. ...

Index - Web APIs
. 50 ambientlightsensor.illuminance api, ambient light level api, generic sensor api, property, reference, sensor, sensor apis, sensors, illuminance the illuminance property of the ambientlightsensor interface returns the current light level in lux of the ambient light level around the hosting device. ...
...ensors, illuminance the illuminance property of the ambientlightsensor interface returns the current light level in lux of the ambient light level around the hosting device. 51 analysernode api, analysernode, interface, 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. 52 analysernode.analysernode() api, analysernode, audio, constructor, media, reference, web audio api the analysernode constructor of the web audio api creates a new analysernode object instance. 53 analyse...
...the output, but allows you to take the generated data, process it, and create audio visualizations. 52 analysernode.analysernode() api, analysernode, audio, constructor, media, reference, web audio api the analysernode constructor of the web audio api creates a new analysernode object instance. 53 analysernode.fftsize api, analysernode, property, reference, web audio api, fftsize the fftsize property of the analysernode interface is an unsigned long value and represents the window size in samples that is used when performing a fast fourier transform (fft) to get frequency domain data. ...
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Media container formats (file types) - Web media technologies
see codecs used by webrtc for information about codecs commonly used for making webrtc calls, as well as browser compatibility information around codec support in webrtc. ...
...see codecs used by webrtc for information about codecs commonly used for making webrtc calls, as well as browser compatibility information around codec support in webrtc. ...
... using mediastreamtrack objects to represent each track. see codecs used by webrtc for information about codecs commonly used for making webrtc calls, as well as browser compatibility information around codec support in webrtc. common container formats while there are a vast number of media container formats, the ones listed below are the ones you are most likely to encounter. some support only audio while others support both audio and video. the mime types and extensions for each are listed.the most commonly used containers for media on the web are probably mpeg-4 (mp4), quicktime movie (mov), and the wavefile audio file format (wav). however, you may also encounter mp3, ogg, webm, avi, and other formats. not all of these are broadly supported by browsers, however; some combinations of conta...
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Introduction to web APIs - Learn web development
art of the javascript language itself, rather they are built on top of the core javascript language, providing you with extra superpowers to use in your javascript code. they generally fall into two categories: browser apis are built into your web browser and are able to expose data from the browser and surrounding computer environment and do useful complex things with it. for example, the web audio api provides javascript constructs for manipulating audio in the browser — taking an audio track, altering its volume, applying effects to it, etc. in the background, the browser is actually using some complex lower-level code (e.g. c++ or rust) to do the actual audio processing. but again, this complexity is abstracted away from you by the api. third-party apis are not built into the browser ...
...draw shapes such as rectangles or circles, import an image onto the canvas, and apply a filter to it such as sepia or grayscale using the canvas api, or create a complex 3d scene with lighting and textures using webgl. such apis are often combined with apis for creating animation loops (such as window.requestanimationframe()) and others to make constantly updating scenes like cartoons and games. ...
... image onto the canvas, and apply a filter to it such as sepia or grayscale using the canvas api, or create a complex 3d scene with lighting and textures using webgl. such apis are often combined with apis for creating animation loops (such as window.requestanimationframe()) and others to make constantly updating scenes like cartoons and games. ...
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Capabilities, constraints, and settings - Web APIs
site or app exchange information about what constrainable properties the browser's implementation supports and what values it supports for each one. this article discusses capabilities and constraints, as well as media settings, and includes an example we call the constraint exerciser. the constraint exerciser lets you experiment with the results of different constraint sets being applied to the audio and video tracks coming from the computer's a/v input devices (such as its webcam and microphone). overview the process works like this (using mediastreamtrack as an example): if needed, call mediadevices.getsupportedconstraints() to get the list of supported constraints, which tells you what constrainable properties the browser knows about. this isn't always necessary, since any that aren't...
...hat you need, and if you can't, decide what to do at that point. see firefox bug 1179084, for example. applying constraints the first and most common way to use constraints is to specify them when you call getusermedia(): navigator.mediadevices.getusermedia({ video: { width: { min: 640, ideal: 1920 }, height: { min: 400, ideal: 1080 }, aspectratio: { ideal: 1.7777777778 } }, audio: { samplesize: 16, channelcount: 2 } }).then(stream => { videoelement.srcobject = stream; }).catch(handleerror); in this example, constraints are applied at getusermedia() time, asking for an ideal set of options with fallbacks for the video. you can specify one or more media input device ids to establish restrictions on which input sources are allowed. to collect a list of the av...
...on whichcamera(track) { return track.getsettings().facingmode; } this function uses getsettings() to obtain the track's currently in-use values for the constrainable properties and returns the value of facingmode. example: constraint exerciser in this example, we create an exerciser which lets you experiment with media constraints by editing the source code describing the constraint sets for audio and video tracks. you can then apply those changes and see the result, including both what the stream looks like and what the actual media settings are set to after applying the new constraints. the html and css for this example are pretty simple, and aren't shown here. you can look at the complete example by clicking here. html content <p>experiment with media constraints! edit the constrain...
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Key Values - Web APIs
with an explanation of what the key is typically used for. corresponding virtual keycodes for common platforms are included where available. learn how to use these key values in javascript using keyboardevent.key special values | modifier keys | whitespace keys | navigation keys | editing keys | ui keys | device keys | ime and composition keys | function keys | phone keys | multimedia keys | audio control keys | tv control keys | media controller keys | speech recognition keys | document keys | application selector keys | browser control keys | numeric keypad keys special values values of key which have special meanings other than identifying a specific key or character. keyboardevent.key value description virtual keycode windows mac linux android ...
...mand_media_channel_down qt::key_channeldown (0x01000119) keycode_channel_down (167) "channelup" switches to the next channel. appcommand_media_channel_up qt::key_channelup (0x01000118) keycode_channel_up (166) "mediafastforward" [2] starts, continues, or increases the speed of fast forwarding the media. ...
...y_channeldown (0x01000119) keycode_channel_down (167) "channelup" switches to the next channel. appcommand_media_channel_up qt::key_channelup (0x01000118) keycode_channel_up (166) "mediafastforward" [2] starts, continues, or increases the speed of fast forwarding the media. ...
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PannerNode - Web APIs
the pannernode interface represents the position and behavior of an audio source signal in space. it is an audionode audio-processing module describing its position with right-hand cartesian coordinates, its movement using a velocity vector and its directionality using a directionality cone. a pannernode always has exactly one input and one output: the input can be mono or stereo but the output is always stereo (2 channels); you can't have panning effects without at least two audio channels! number of inputs 1 number of outputs 1 channel count mode "clamped-max" channel count 2 channel interpretation "speakers" constructor pannernode.pannernode creates a new pannernode object instance. properties inherits propertie...
...retrieval is done by accessing, for example, pannernode.positionx. while setting the same property is done with pannernode.positionx.value. ...
...retrieval is done by accessing, for example, pannernode.positionx. ...
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The "codecs" parameter in common media types - Web media technologies
at a fundamental level, you can specify the type of a media file using a simple mime type, such as video/mp4 or audio/mpeg. however, many media types—especially those that support video tracks—can benefit from the ability to more precisely describe the format of the data within them. for instance, just describing a video in an mpeg-4 file with the mime type video/mp4 doesn't say anything about what format the actual media within takes. for that reason, the codecs parameter can be added to the mime type describing media content. with it, container-specific information can be provided. this information may include things like the profile of the video codec, the type used for the audio tracks, and so forth. this guide briefly examines the syntax of the media type codecs parameter and how it...
...ter can be added to the mime type describing media content. with it, container-specific information can be provided. this information may include things like the profile of the video codec, the type used for the audio tracks, and so forth. this guide briefly examines the syntax of the media type codecs parameter and how it's used with the mime type string to provide details about the contents of audio or video media beyond simply indicating the container type. general syntax a basic mime media type is expressed by stating the type of media (audio, video, etc), then a slash character (/), then the container format used to contain the media: audio/mpeg an audio file using the mpeg file type, such as an mp3. video/ogg a video file using the ogg file type. video/mp4 a video file using th...
...eason, you can add the codecs parameter to the media type. to do so, append a semicolon (;) followed by codecs= and then the string describing the format of the contents of the file. some media types only let you specify the names of the codecs to use, while others allow you to specify various constraints on those codecs as well. you can specify multiple codecs by separating them with commas. audio/ogg; codecs=vorbis an ogg file containing a vorbis audio track. video/webm; codecs="vp8, vorbis" a webm file containing vp8 video and/or vorbis audio. video/mp4; codecs="avc1.4d002a" an mpeg-4 file containing avc (h.264) video, main profile, level 4.2. as is the case with any mime type parameter, codecs must be changed to codecs* (note the asterisk character, *) if any of the properties of...
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Using DTMF with WebRTC - Web APIs
instead, they're sent out-of-band, as rtp payloads. note, however, that although it's possible to send dtmf using webrtc, there is currently no way to detect or receive incoming dtmf. ...
...io/video conferencing, webrtc supports sending dtmf to the remote peer on an rtcpeerconnection. this article offers a brief high-level overview of how dtmf works over webrtc, then provides a guide for everyday developers about how to send dtmf over an rtcpeerconnection. the dtmf system is often referred to as "touch tone," after an old trade name for the system. webrtc doesn't send dtmf codes as audio data. instead, they're sent out-of-band, as rtp payloads. note, however, that although it's possible to send dtmf using webrtc, there is currently no way to detect or receive incoming dtmf. webrtc currently ignores these payloads; this is because webrtc's dtmf support is primarily intended for use with legacy telephone services that rely on dtmf tones to perform tasks such as: teleconferencing...
... receive incoming dtmf. webrtc currently ignores these payloads; this is because webrtc's dtmf support is primarily intended for use with legacy telephone services that rely on dtmf tones to perform tasks such as: teleconferencing systems menu systems voicemail systems entry of credit card or other payment information passcode entry note: while the dtmf is not sent to the remote peer as audio, browsers may choose to play the corresponding tone to the local user as part of their user experience, since users are typically used to hearing their phone play the tones audibly. sending dtmf on an rtcpeerconnection a given rtcpeerconnection can have multiple media tracks sent or received on it. when you wish to transmit dtmf signals, you first need to decide which track to send them on, si...
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Accessible multimedia - Learn web development
this article shows how. prerequisites: basic computer literacy, a basic understanding of html, css, and javascript, an understanding of what accessibility is. objective: to understand the accessibility issues behind multimedia, and how to overcome them. ...
... can refer back to there for the full details. in short, you should ensure that where possible visual content has an alternative text available for screenreaders to pick up and read to their users. ...
...let's investigate. ...
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Codecs used by WebRTC - Web media technologies
to communicate, the two devices need to be able to agree upon a mutually-understood codec for each track so they can successfully communicate and present the shared media. this guide reviews the codecs that browsers are required to implement as well as other codecs that some or all browsers support for webrtc. containerless media webrtc uses bare mediastreamtrack objects for each track being shared from one peer to another, without a container or even a mediastream associated with the tracks. ...
...codecs that browsers are required to implement as well as other codecs that some or all browsers support for webrtc. while compression is always a necessity when dealing with media on the web, it's of additional importance when videoconferencing in order to ensure that the participants are able to communicate without lag or interruptions. of secondary importance is the need to keep the video and audio synchronized, so that the movements and any ancillary information (such as slides or a projection) are presented at the same time as the audio that corresponds. general codec requirements before looking at codec-specific capabilities and requirements, there are a few overall requirements that must be met by any codec configuration used with webrtc. unless the sdp specifically signals otherwise...
...wise, the pixel aspect ratio is 1:1, indicating that pixels are square. other notes the payload format used for avc in webrtc is described in rfc 6184: rtp payload format for h.264 video. avc implementations for webrtc are required to support the special "filler payload" and "full frame freeze" sei messages; these are used to support switching among multiple input streams seamlessly. ...
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AnalyserNode.getFloatFrequencyData() - Web APIs
y represents the decibel value for a specific frequency. the frequencies are spread linearly from 0 to 1/2 of the sample rate. for example, for 48000 sample rate, the last item of the array will represent the decibel value for 24000 hz. if you need higher performance and don't care about precision, you can use analysernode.getbytefrequencydata() instead, which works on a uint8array. ...
...e decibel value for a specific frequency. the frequencies are spread linearly from 0 to 1/2 of the sample rate. for example, for 48000 sample rate, the last item of the array will represent the decibel value for 24000 hz. if you need higher performance and don't care about precision, you can use analysernode.getbytefrequencydata() instead, which works on a uint8array. syntax var audioctx = new audiocontext(); var analyser = audioctx.createanalyser(); var dataarray = new float32array(analyser.frequencybincount); // float32array should be the same length as the frequencybincount void analyser.getfloatfrequencydata(dataarray); // fill the float32array with data returned from getfloatfrequencydata() parameters array the float32array that the frequency domain data will be copied to. for an...
...ith data returned from getfloatfrequencydata() parameters array the float32array that the frequency domain data will be copied to. for any sample which is silent, the value is -infinity. if the array has fewer elements than the analysernode.frequencybincount, excess elements are dropped. if it has more elements than needed, excess elements are ignored. return value none. ...
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Media buffering, seeking, and time ranges - Developer guides
sometimes it's useful to know how much <audio> or <video> has downloaded or is playable without delay — a good example of this is the buffered progress bar of an audio or video player. this article discusses how to build a buffer/seek bar using timeranges, and other features of the media api. buffered the buffered attribute will tell us which parts of the media has been downloaded. it returns a timeranges object, which will tell us which chunks of media have been downloaded. this is usually contiguous but if the user jumps about while media is buffering, it may contain holes. ...
...how to build a buffer/seek bar using timeranges, and other features of the media api. buffered the buffered attribute will tell us which parts of the media has been downloaded. it returns a timeranges object, which will tell us which chunks of media have been downloaded. this is usually contiguous but if the user jumps about while media is buffering, it may contain holes. this will work with <audio> or <video>; for now let's consider a simple audio example: <audio id="my-audio" controls src="music.mp3"> </audio> we can access these attributes like so: var myaudio = document.getelementbyid('my-audio'); var bufferedtimeranges = myaudio.buffered; timeranges object timeranges are a series of non-overlapping ranges of time, with start and stop times. (learn more about timeranges). a timer...
...n appear, as illustrated by the below visualization. this represents two buffered time ranges — one spanning 0 to 5 seconds and the second spanning 15 to 19 seconds. ...
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BiquadFilterNode - Web APIs
the biquadfilternode interface represents a simple low-order filter, and is created using the audiocontext.createbiquadfilter() method. it is an audionode that can represent different kinds of filters, tone control devices, and graphic equalizers. a biquadfilternode always has exactly one input and one output. number of inputs 1 number of outputs 1 channel count mode "max" channel count 2 (not used in the default count mode) channel interpretation "speakers" constructor biquadfilternode() creates a new instance of a biquadfilternode object. properties inherits properties from its parent, audionode. note: though the audioparam objects returned are read-only, the values they represent are not. biquadfilternode.freq...
...nt 2 (not used in the default count mode) channel interpretation "speakers" constructor biquadfilternode() creates a new instance of a biquadfilternode object. properties inherits properties from its parent, audionode. note: though the audioparam objects returned are read-only, the values they represent are not. biquadfilternode.frequency read only is an a-rate audioparam, a double representing a frequency in the current filtering algorithm measured in hertz (hz). ...
... of a biquadfilternode object. properties inherits properties from its parent, audionode. note: though the audioparam objects returned are read-only, the values they represent are not. biquadfilternode.frequency read only is an a-rate audioparam, a double representing a frequency in the current filtering algorithm measured in hertz (hz). biquadfilternode.detune read only is an a-rate audioparam representing detuning of the frequency in cents. biquadfilternode.q read only is an a-rate audioparam, a double representing a q factor, or quality factor. biquadfilternode.gain read only is an a-rate audioparam, a double representing the gain used in the current filtering algorithm. biquadfilternode.type is a string value defining the kind of filtering algorithm the node is implemen...
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PannerNode.distanceModel - Web APIs
the distancemodel property of the pannernode interface is an enumerated value determining which algorithm to use to reduce the volume of the audio source as it moves away from the listener. the possible values are: linear: a linear distance model calculating the gain induced by the distance according to: 1 - rollofffactor * (distance - refdistance) / (maxdistance - refdistance) inverse: an inverse distance model calculating the gain induced by the distance according to: refdistance / (refdistance + rollofffactor * (math.max(distance, refdistance) - refdistance)) exponential: an exponential distance model calculating the gain induced by the distance according to: pow((math.max(distance, refdistance) / refdistance, -rollofffactor). inverse is the default value of distancemodel. ...
... model calculating the gain induced by the distance according to: refdistance / (refdistance + rollofffactor * (math.max(distance, refdistance) - refdistance)) exponential: an exponential distance model calculating the gain induced by the distance according to: pow((math.max(distance, refdistance) / refdistance, -rollofffactor). inverse is the default value of distancemodel. ...
...ting the gain induced by the distance according to: refdistance / (refdistance + rollofffactor * (math.max(distance, refdistance) - refdistance)) exponential: an exponential distance model calculating the gain induced by the distance according to: pow((math.max(distance, refdistance) / refdistance, -rollofffactor). inverse is the default value of distancemodel. ...
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Introduction to the Real-time Transport Protocol (RTP) - Web APIs
rantee ordering or even guarantee delivery at all. this means rtp can be — but is not required to be — used atop udp for its performance as well as its multiplexing and checksum features. rtp supports multicast; while this isn't yet important for webrtc, it's likely to matter in the future, when webrtc is (hopefully) enhanced to support multi-user conversations. rtp isn't limited to use in audiovisual communication. it can be used for any form of continuous or active data transfer, including data streaming, active badges or status display updates, or control and measurement information transport. things rtp doesn't do rtp itself doesn't provide every possible feature, which is why other protocols are also used by webrtc. some of the more noteworthy things rtp doesn't include: editor'...
... rtcpeerconnection are implemented using rtp and the interfaces above, you can take advantage of the access this gives you to the internals of streams to make adjustments. among the simplest things you can do is to implement a "hold" feature, wherein a participant in a call can click a button and turn off their microphone, begin sending music to the other peer instead, and stop accepting incoming audio. note: this example makes use of modern javascript features including async functions and the await expression. this enormously simplifies and makes far more readable the code dealing with the promises returned by webrtc methods. in the examples below, we'll refer to the peer which is turning "hold" mode on and off as the local peer and the user being placed on hold as the remote peer. activ...
...he code dealing with the promises returned by webrtc methods. in the examples below, we'll refer to the peer which is turning "hold" mode on and off as the local peer and the user being placed on hold as the remote peer. activating hold mode local peer when the local user decides to enable hold mode, the enablehold() method below is called. ...
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HTMLMediaElement - Web APIs
the htmlmediaelement interface adds to htmlelement the properties and methods needed to support basic media-related capabilities that are common to audio and video. the htmlvideoelement and htmlaudioelement elements both inherit this interface. <div id="interfacediagram" style="display: inline-block; position: relative; width: 100%; padding-bottom: 30%; vertical-align: middle; overflow: hidden;"><svg style="display: inline-block; position: absolute; top: 0; left: 0;" viewbox="-50 0 600 180" preserveaspectratio="xminymin meet"><a xlink:href="/docs/web/api/eventtarget" target="_top"><rect x="1" y="1" width="110" height="50" fill="#fff" stroke="#d4dde4" stroke-width="2px" /><text x="56" y="30" font-size="12px" font-family="consolas,monaco,andale mono,monospace" fill="#4d4e53" text-anchor="m...
...2px" /><text x="411" y="94" font-size="12px" font-family="consolas,monaco,andale mono,monospace" fill="#4d4e53" text-anchor="middle" alignment-baseline="middle">htmlmediaelement</text></a></svg></div> a:hover text { fill: #0095dd; pointer-events: all;} properties this interface also inherits properties from its ancestors htmlelement, element, node, and eventtarget. ...
...="411" y="94" font-size="12px" font-family="consolas,monaco,andale mono,monospace" fill="#4d4e53" text-anchor="middle" alignment-baseline="middle">htmlmediaelement</text></a></svg></div> a:hover text { fill: #0095dd; pointer-events: all;} properties this interface also inherits properties from its ancestors htmlelement, element, node, and eventtarget. ...
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<video>: The Video Embed element - HTML: Hypertext Markup Language
the source for this interactive example is stored in a github repository. if you'd like to contribute to the interactive examples project, please clone https://github.com/mdn/interactive-examples and send us a pull request. the above example shows simple usage of the <video> element. ...
...e the opening and closing <video></video> tags is shown as a fallback in browsers that don't support the element. attributes like all other html elements, this element supports the global attributes. autoplay a boolean attribute; if specified, the video automatically begins to play back as soon as it can do so without stopping to finish loading the data. note: sites that automatically play audio (or videos with an audio track) can be an unpleasant experience for users, so should be avoided when possible. if you must offer autoplay functionality, you should make it opt-in (requiring a user to specifically enable it). however, this can be useful when creating media elements whose source will be set at a later time, under user control. see our autoplay guide for additional information about...
...pecified in the attribute. specifically, the image would raster at these dimensions and naturalwidth/naturalheight on images would return the values specified in this attribute. explainer, examples loop a boolean attribute; if specified, the browser will automatically seek back to the start upon reaching the end of the video. muted a boolean attribute that indicates the default setting of the audio contained in the video. if set, the audio will be initially silenced. its default value is false, meaning that the audio will be played when the video is played. playsinline a boolean attribute indicating that the video is to be played "inline", that is within the element's playback area. note that the absence of this attribute does not imply that the video will always be played in fullscreen. ...
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DelayNode.delayTime - Web APIs
example we have created a simple example that allows you to play three different samples on a constant loop — see create-delay (you can also view the source code). ...
... example we have created a simple example that allows you to play three different samples on a constant loop — see create-delay (you can also view the source code). ...
... example we have created a simple example that allows you to play three different samples on a constant loop — see create-delay (you can also view the source code). ...
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PannerNode.setOrientation() - Web APIs
the setorientation() method of the pannernode interface defines the direction the audio source is playing in. this can have a big effect if the sound is very directional — controlled by the three cone-related attributes pannernode.coneinnerangle, pannernode.coneouterangle, and pannernode.coneoutergain. in such a case, a sound pointing away from the listener can be very quiet or even silent. the three parameters x, y and z are unitless and describe a direction vector in 3d space using the right-hand cartesian coordinate system. the default value of the direction vector is (1, 0, 0). syntax var audioctx = new audiocontext(); var panner = audioctx.createpanner(); panner.setorientation(1,0,0); returns void. example in the following example, you can see an example of how the createpanne...
...ers x, y and z are unitless and describe a direction vector in 3d space using the right-hand cartesian coordinate system. the default value of the direction vector is (1, 0, 0). syntax var audioctx = new audiocontext(); var panner = audioctx.createpanner(); panner.setorientation(1,0,0); returns void. example in the following example, you can see an example of how the createpanner() method, audiolistener and pannernode would be used to control audio spatialisation. ...
...e of the direction vector is (1, 0, 0). syntax var audioctx = new audiocontext(); var panner = audioctx.createpanner(); panner.setorientation(1,0,0); returns void. example in the following example, you can see an example of how the createpanner() method, audiolistener and pannernode would be used to control audio spatialisation. generally you will define the position in 3d space that your audio listener and panner (source) occupy initially, and then update the position of one or both of these as the application is used. you might be moving a character around inside a game world for example, and wanting delivery of audio to change realistically as your character moves closer to or further away from a music player such as a stereo. in the example you can see this being controlled by the f...
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ScriptProcessorNode - Web APIs
the scriptprocessornode interface allows the generation, processing, or analyzing of audio using javascript. note: as of the august 29 2014 web audio api spec publication, this feature has been marked as deprecated, and was replaced by audioworklet (see audioworkletnode). the scriptprocessornode interface is an audionode audio-processing module that is linked to two buffers, one containing the input audio data, one containing the processed output audio data. an event, implementing the audioprocessingevent interface, is sent to the object each time the input buffer contains new data, and the event handler terminates when it has filled the output buffer with data. ...
...nked to two buffers, one containing the input audio data, one containing the processed output audio data. an event, implementing the audioprocessingevent interface, is sent to the object each time the input buffer contains new data, and the event handler terminates when it has filled the output buffer with data. the size of the input and output buffer are defined at the creation time, when the audiocontext.createscriptprocessor() method is called (both are defined by audiocontext.createscriptprocessor()'s buffersize parameter). the buffer size must be a power of 2 between 256 and 16384, that is 256, 512, 1024, 2048, 4096, 8192 or 16384. small numbers lower the latency, but large number may be necessary to avoid audio breakup and glitches. if the buffer size is not defined, which is recommen...
...glitches. if the buffer size is not defined, which is recommended, the browser will pick one that its heuristic deems appropriate. number of inputs 1 number of outputs 1 channel count mode "max" channel count 2 (not used in the default count mode) channel interpretation "speakers" properties inherits properties from its parent, audionode. scriptprocessornode.buffersize read only returns an integer representing both the input and output buffer size. its value can be a power of 2 value in the range 256–16384. methods no specific methods; inherits methods from its parent, audionode. events listen to these events using addeventlistener() or by assigning an event listener to the oneventname property of this interface: ...
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StereoPannerNode - Web APIs
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. ...
... 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. ...
... apply a simple panning effect than having to use a full pannernode. number of inputs 1 number of outputs 1 channel count mode "clamped-max" channel count 2 channel interpretation "speakers" constructor stereopannernode() creates a new instance of a stereopannernode object. ...
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BiquadFilterNode.type - Web APIs
type values and their meaning type description frequency q gain lowpass standard second-order resonant lowpass filter with 12db/octave rolloff. frequencies below the cutoff pass through; frequencies above it are attenuated. the cutoff frequency. ...
... changes the phase-relationship between the various frequencies. the frequency with the maximal group delay, that is, the frequency where the center of the phase transition occurs. controls how sharp the transition is at the medium frequency. the larger this parameter is, the sharper and larger the transition will be. ...
...f the phase transition occurs. controls how sharp the transition is at the medium frequency. the larger this parameter is, the sharper and larger the transition will be. not used example the following example shows basic usage of an audiocontext to create a biquad filter node. for a complete working example, check out our voice-change-o-matic demo (look at the source code too). ...
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DynamicsCompressorNode - Web APIs
the dynamicscompressornode interface provides a compression effect, which lowers the volume of the loudest parts of the signal in order to help prevent clipping and distortion that can occur when multiple sounds are played and multiplexed together at once. this is often used in musical production and game audio. dynamicscompressornode is an audionode that has exactly one input and one output; it is created using the audiocontext.createdynamicscompressor() method. number of inputs 1 number of outputs 1 channel count mode "clamped-max" channel count 2 channel interpretation "speakers" constructor dynamicscompressornode() creates a new instance of an dynamicscompressornode object. properties inherits properties from its ...
...sing the audiocontext.createdynamicscompressor() method. number of inputs 1 number of outputs 1 channel count mode "clamped-max" channel count 2 channel interpretation "speakers" constructor dynamicscompressornode() creates a new instance of an dynamicscompressornode object. properties inherits properties from its parent, audionode. dynamicscompressornode.threshold read only is a k-rate audioparam representing the decibel value above which the compression will start taking effect. dynamicscompressornode.knee read only is a k-rate audioparam containing a decibel value representing the range above the threshold where the curve smoothly transitions to the compressed portion. dynamicscompressornode.ratio read only ...
...icscompressornode.threshold read only is a k-rate audioparam representing the decibel value above which the compression will start taking effect. dynamicscompressornode.knee read only is a k-rate audioparam containing a decibel value representing the range above the threshold where the curve smoothly transitions to the compressed portion. dynamicscompressornode.ratio read only is a k-rate audioparam representing the amount of change, in db, needed in the input for a 1 db change in the output. dynamicscompressornode.reduction read only is a float representing the amount of gain reduction currently applied by the compressor to the signal. dynamicscompressornode.attack read only is a k-rate audioparam representing the amount of time, in seconds, required to reduce the gain by 10 db. ...
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PannerNode.maxDistance - Web APIs
the maxdistance property of the pannernode interface is a double value representing the maximum distance between the audio source and the listener, after which the volume is not reduced any further. this value is used only by the linear distance model. the maxdistance property's default value is 10000. syntax var audioctx = new audiocontext(); var panner = audioctx.createpanner(); panner.maxdistance = 10000; value a double. the default is 10000, and non-positive values are not allowed. exceptions rangeerror the property has been given a value that is outside the accepted range. example in the following example, you can see an example of how the createpanner() method, audiolistener and pannernode would be used to control audio spatialisation. generally you will define the positi...
...uble. the default is 10000, and non-positive values are not allowed. exceptions rangeerror the property has been given a value that is outside the accepted range. example in the following example, you can see an example of how the createpanner() method, audiolistener and pannernode would be used to control audio spatialisation. generally you will define the position in 3d space that your audio listener and panner (source) occupy initially, and then update the position of one or both of these as the application is used. you might be moving a character around inside a game world for example, and wanting delivery of audio to change realistically as your character moves closer to or further away from a music player such as a stereo. in the example you can see this being controlled by the f...
...tion via the positionpanner() function. to see a complete implementation, check out our panner-node example (view the source code) — this demo transports you to the 2.5d "room of metal", where you can play a track on a boom box and then walk around the boom box to see how the sound changes! note how we have used some feature detection to either give the browser the newer property values (like audiolistener.forwardx) for setting position, etc. if it supports those, or older methods (like audiolistener.setorientation()) if it still supports those but not the new properties. // set up listener and panner position information var width = window.innerwidth; var height = window.innerheight; var xpos = math.floor(width/2); var ypos = math.floor(height/2); var zpos = 295; // define other variabl...
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PannerNode.panningModel - Web APIs
the panningmodel property of the pannernode interface is an enumerated value determining which spatialisation algorithm to use to position the audio in 3d space. the possible values are: equalpower: represents the equal-power panning algorithm, generally regarded as simple and efficient. equalpower is the default value. hrtf: renders a stereo output of higher quality than equalpower — it uses a convolution with measured impulse responses from human subjects. syntax var audioctx = new audiocontext(); var panner = audioctx.createpanner(); panner.panningmodel = 'hrtf'; value a enum — see panningmodeltype. example in the following example, you can see an example of how the createpanner() method, audiolistener and pannernode would be used to control audio spatialisation. generall...
...man subjects. syntax var audioctx = new audiocontext(); var panner = audioctx.createpanner(); panner.panningmodel = 'hrtf'; value a enum — see panningmodeltype. example in the following example, you can see an example of how the createpanner() method, audiolistener and pannernode would be used to control audio spatialisation. generally you will define the position in 3d space that your audio listener and panner (source) occupy initially, and then update the position of one or both of these as the application is used. you might be moving a character around inside a game world for example, and wanting delivery of audio to change realistically as your character moves closer to or further away from a music player such as a stereo. in the example you can see this being controlled by the f...
...tion via the positionpanner() function. to see a complete implementation, check out our panner-node example (view the source code) — this demo transports you to the 2.5d "room of metal", where you can play a track on a boom box and then walk around the boom box to see how the sound changes! note how we have used some feature detection to either give the browser the newer property values (like audiolistener.forwardx) for setting position, etc. if it supports those, or older methods (like audiolistener.setorientation()) if it still supports those but not the new properties. // set up listener and panner position information var width = window.innerwidth; var height = window.innerheight; var xpos = math.floor(width/2); var ypos = math.floor(height/2); var zpos = 295; // define other variabl...
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PannerNode.setPosition() - Web APIs
the setposition() method of the pannernode interface defines the position of the audio source relative to the listener (represented by an audiolistener object stored in the audiocontext.listener attribute.) the three parameters x, y and z are unitless and describe the source's position in 3d space using the right-hand cartesian coordinate system. the setposition() method's default value of the position is (0, 0, 0). syntax var audioctx = new audiocontext(); var panner = audioctx.createpanner(); panner.setposition(0,0,0); returns void. example in the following example, you can see an example of how the createpanner() method, audiolistener and pannernode would be used to control audio spatialisation. ...
...efault value of the position is (0, 0, 0). syntax var audioctx = new audiocontext(); var panner = audioctx.createpanner(); panner.setposition(0,0,0); returns void. example in the following example, you can see an example of how the createpanner() method, audiolistener and pannernode would be used to control audio spatialisation. generally you will define the position in 3d space that your audio listener and panner (source) occupy initially, and then update the position of one or both of these as the application is used. you might be moving a character around inside a game world for example, and wanting delivery of audio to change realistically as your character moves closer to or further away from a music player such as a stereo. in the example you can see this being controlled by the f...
...tion via the positionpanner() function. to see a complete implementation, check out our panner-node example (view the source code) — this demo transports you to the 2.5d "room of metal", where you can play a track on a boom box and then walk around the boom box to see how the sound changes! note how we have used some feature detection to either give the browser the newer property values (like audiolistener.forwardx) for setting position, etc. if it supports those, or older methods (like audiolistener.setorientation()) if it still supports those but not the new properties. // set up listener and panner position information var width = window.innerwidth; var height = window.innerheight; var xpos = math.floor(width/2); var ypos = math.floor(height/2); var zpos = 295; // define other variabl...
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PannerNode.setVelocity() - Web APIs
the setvelocity() method of the pannernode interface defines the velocity vector of the audio source — how fast it is moving and in what direction. this method was removed from the specification because of gaps in its design and implementation problems. the velocity relative to the listener is used to control the pitch change needed to conform with the doppler effect due to the relative speed. as the vector controls both the direction of travel and its velocity, the three parameters x, y and z are expressed in meters per second. the default value of the velocity vector is (0, 0, 0). syntax var audioctx = new audiocontext(); var panner = audioctx.createpanner(); panner.setvelocity(0,0,17); returns void. example in the following example, you can see an example of how the createpanne...
...r controls both the direction of travel and its velocity, the three parameters x, y and z are expressed in meters per second. the default value of the velocity vector is (0, 0, 0). syntax var audioctx = new audiocontext(); var panner = audioctx.createpanner(); panner.setvelocity(0,0,17); returns void. example in the following example, you can see an example of how the createpanner() method, audiolistener and pannernode would be used to control audio spatialisation. ...
...value of the velocity vector is (0, 0, 0). syntax var audioctx = new audiocontext(); var panner = audioctx.createpanner(); panner.setvelocity(0,0,17); returns void. example in the following example, you can see an example of how the createpanner() method, audiolistener and pannernode would be used to control audio spatialisation. generally you will define the position in 3d space that your audio listener and panner (source) occupy initially, and then update the position of one or both of these as the application is used. you might be moving a character around inside a game world for example, and wanting delivery of audio to change realistically as your character moves closer to or further away from a music player such as a stereo. in the example you can see this being controlled by the f...
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Perceivable - Accessibility
omponents must be presentable to users in ways they can perceive. guideline 1.1 — providing text alternatives for non-text content the key here is that text can be converted to other forms that people with disabilities can use. for example, it can be spoken by a screenreader, converted to large print, or represented on a braille display. non-text content refers to multimedia such as images, audio, and video. success criteria how to conform to the criteria practical resource 1.1.1 provide text equivalents (a) all images that convey meaningful content should be given suitable alternative text. text alternatives. complex images or charts should have an accessible alternative provided, either on the same page or via a link. use a regular link inste...
... complex images or charts should have an accessible alternative provided, either on the same page or via a link. use a regular link instead of a longdesc attribute. a text description may work, or an accessible data table (see html table advanced features and accessibility). see other text alternative mechanisms for the argument against longdesc. multimedia content (i.e., audio or video) should at least have a descriptive identification available, such as a caption or similar. see text alternatives for static caption options, and audio transcripts, video text tracks, and other multimedia content for other alternatives. ui controls such as form elements and buttons should have text labels that describe their purpose. buttons are simple—you shoul...
...ontent) images, video, etc., in a way that is invisible to assistive technology, so it doesn't confuse users. decorative images should be implemented using css background images (see backgrounds). if you have to include an image via an <img> element, give it a blank alt (alt=""). otherwise, screenreaders may try to read out the filepath, etc. if you are including background video or audio that autoplays, make it as unobtrusive as possible. don't make it look/sound like content, and provide a control to turn it off. ideally, don't include it at all. note: also see the wcag description for guideline 1.1: text alternatives. guideline 1.2 — providing text alternatives for time-based media time-based media refers to multimedia with a duration, such as audio and video. ...
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Index - Developer guides
getting started ajax, api, advanced, javascript, webmechanics, xmlhttprequest this article guides you through the ajax basics and gives you some simple hands-on examples to get you started. 5 wai aria live regions/api support ajax, accessibility firefox 3 contains important improvements to the way the mozilla engine exposes live changes in a document. 6 audio and video delivery audio, guide, html, html5, media, video whether we are dealing with pre-recorded audio files or live streams, the mechanism for making them available through the browser's <audio> and <video> elements remains pretty much the same. currently, to support all browsers we need to specify two formats, although with the adoption of mp3 and mp4 formats in firefox and opera, ...
...hat uses the media and fullscreen apis and works across most major desktop and mobile browsers. as well as working fullscreen, the player features custom controls rather than just using the browser defaults. the player controls themselves won't be styled beyond the basics required to get them working; full styling of the player will be taken care of in a future article. ...
...fullscreen apis and works across most major desktop and mobile browsers. as well as working fullscreen, the player features custom controls rather than just using the browser defaults. the player controls themselves won't be styled beyond the basics required to get them working; full styling of the player will be taken care of in a future article. ...
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BiquadFilterNode.gain - Web APIs
the gain property of the biquadfilternode interface is a a-rate audioparam, a double representing the gain used in the current filtering algorithm. when it takes a positive value it is a real gain, when negative it is an attenuation. it is expressed in db, has a default value of 0 and can take a value in a nominal range of -40 to 40. syntax var audioctx = new audiocontext(); var biquadfilter = audioctx.createbiquadfilter(); biquadfilter.gain.value = 25; note: though the audioparam returned is read-only, the value it represents is not. value an audioparam. example the following example shows basic usage of an audiocontext to create a biquad filter node. ...
...; var biquadfilter = audioctx.createbiquadfilter(); biquadfilter.gain.value = 25; note: though the audioparam returned is read-only, the value it represents is not. value an audioparam. example the following example shows basic usage of an audiocontext to create a biquad filter node. for a complete working example, check out our voice-change-o-matic demo (look at the source code too). ...
...dioctx.createbiquadfilter(); biquadfilter.gain.value = 25; note: though the audioparam returned is read-only, the value it represents is not. value an audioparam. example the following example shows basic usage of an audiocontext to create a biquad filter node. for a complete working example, check out our voice-change-o-matic demo (look at the source code too). ...
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Controlling multiple parameters with ConstantSourceNode - Web APIs
this article demonstrates how to use a constantsourcenode to link multiple parameters together so they share the same value, which can be changed by simply setting the value of the constantsourcenode.offset parameter. you may have times when you want to have multiple audio parameters be linked so they share the same value even while being changed in some way. ...
... you may have times when you want to have multiple audio parameters be linked so they share the same value even while being changed in some way. for example, perhaps you have a set of oscillators, and two of them need to share the same, configurable volume, or you have a filter that's been applied to certain inputs but not to all of them. ...
...pplied to certain inputs but not to all of them. you could use a loop and change the value of each affected audioparam one at a time, but there are two drawbacks to doing it that way: first, that's extra code that, as you're about to see, you don't have to write; and second, that loop uses valuable cpu time on your thread (likely the main thread), and there's a way to offload all that work to the audio rendering thread, which is optimized for this kind of work and may run at a more appropriate priority level than your code. the solution is simple, and it involves using an audio node type which, at first glance, doesn't look all that useful: constantsourcenode. the technique this is actually a really easy way to do something that sounds like it might be hard to do. you simply need to create a...
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Using IIR filters - Web APIs
you can check out the full demo here on codepen. ...
... you can check out the full demo here on codepen. ...
...ts a general infinite impulse response (iir) filter; this type of filter can be used to implement tone control devices and graphic equalizers, and the filter response parameters can be specified, so that it can be tuned as needed. this article looks at how to implement one, and use it in a simple example. demo our simple example for this guide provides a play/pause button that starts and pauses audio play, and a toggle that turns an iir filter on and off, altering the tone of the sound. it also provides a canvas on which is drawn the frequency response of the audio, so you can see what effect the iir filter has. you can check out the full demo here on codepen. also see the source code on github. it includes some different coefficient values for different lowpass frequencies — you can cha...
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BiquadFilterNode.Q - Web APIs
the q property of the biquadfilternode interface is an a-rate audioparam, a double representing a q factor, or quality factor. it is a dimensionless value with a default value of 1 and a nominal range of 0.0001 to 1000. syntax var audioctx = new audiocontext(); var biquadfilter = audioctx.createbiquadfilter(); biquadfilter.q.value = 100; note: though the audioparam returned is read-only, the value it represents is not. value an audioparam. example the following example shows basic usage of an audiocontext to create a biquad filter node. for a complete working example, check out our voice-change-o-matic demo (look at the source code too). ...
...(); var biquadfilter = audioctx.createbiquadfilter(); biquadfilter.q.value = 100; note: though the audioparam returned is read-only, the value it represents is not. value an audioparam. example the following example shows basic usage of an audiocontext to create a biquad filter node. for a complete working example, check out our voice-change-o-matic demo (look at the source code too). ...
...audioctx.createbiquadfilter(); biquadfilter.q.value = 100; note: though the audioparam returned is read-only, the value it represents is not. value an audioparam. example the following example shows basic usage of an audiocontext to create a biquad filter node. for a complete working example, check out our voice-change-o-matic demo (look at the source code too). ...
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BiquadFilterNode.detune - Web APIs
the detune property of the biquadfilternode interface is an a-rate audioparam representing detuning of the frequency in cents. syntax var audioctx = new audiocontext(); var biquadfilter = audioctx.createbiquadfilter(); biquadfilter.detune.value = 100; note: though the audioparam returned is read-only, the value it represents is not. value an a-rate audioparam. example the following example shows basic usage of an audiocontext to create a biquad filter node. for a complete working example, check out our voice-change-o-matic demo (look at the source code too). ...
...adfilter = audioctx.createbiquadfilter(); biquadfilter.detune.value = 100; note: though the audioparam returned is read-only, the value it represents is not. value an a-rate audioparam. example the following example shows basic usage of an audiocontext to create a biquad filter node. for a complete working example, check out our voice-change-o-matic demo (look at the source code too). ...
...atebiquadfilter(); biquadfilter.detune.value = 100; note: though the audioparam returned is read-only, the value it represents is not. value an a-rate audioparam. example the following example shows basic usage of an audiocontext to create a biquad filter node. for a complete working example, check out our voice-change-o-matic demo (look at the source code too). ...
...And 9 more matches

BiquadFilterNode.frequency - Web APIs
the frequency property of the biquadfilternode interface is a k-rate audioparam, a double representing a frequency in the current filtering algorithm measured in hertz (hz). frequency's default value is 350 with a nominal range of 10 to the nyquist frequency — that is, half of the sample rate. syntax var audioctx = new audiocontext(); var biquadfilter = audioctx.createbiquadfilter(); biquadfilter.frequency.value = 3000; note: though the audioparam returned is read-only, the value it represents is not. value an audioparam. example the following example shows basic usage of an audiocontext to create a biquad filter node. ...
...iquadfilter = audioctx.createbiquadfilter(); biquadfilter.frequency.value = 3000; note: though the audioparam returned is read-only, the value it represents is not. value an audioparam. example the following example shows basic usage of an audiocontext to create a biquad filter node. for a complete working example, check out our voice-change-o-matic demo (look at the source code too). ...
...createbiquadfilter(); biquadfilter.frequency.value = 3000; note: though the audioparam returned is read-only, the value it represents is not. value an audioparam. example the following example shows basic usage of an audiocontext to create a biquad filter node. for a complete working example, check out our voice-change-o-matic demo (look at the source code too). ...
...And 9 more matches

Using the MediaStream Recording API - Web APIs
the mediastream recording api makes it easy to record audio and/or video streams. when used with navigator.mediadevices.getusermedia(), it provides an easy way to record from the user's input devices and instantly use the result in web apps. both audio and video may be recorded, separately or together. this article aims to provide a basic guide on how to use the mediarecorder interface, which provides this api. a sample application: web dictaphone to demonstrate basic usage of the mediarecorder api, we have built a web-based dictaphone. it allows you to record snippets of audio and then play them back. it even gives you a visualization of your device's sound input, using the web audio api. we'll concentrate on the recording and playback functionality for this article. you can see this d...
...label), the adjacent <aside> element will have its horizontal translation value changed and transition smoothly into view: input[type=checkbox]:checked ~ aside { transform: translatex(0); } basic app setup to grab the media stream we want to capture, we use getusermedia(). ...
... aside { transform: translatex(0); } basic app setup to grab the media stream we want to capture, we use getusermedia(). we then use the mediarecorder api to record the stream, and output each recorded snippet into the source of a generated <audio> element so it can be played back. ...
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MIME types (IANA media types) - HTTP
binary data that doesn't fall explicitly into one of the other types; either data that will be executed or interpreted in some way or binary data that requires a specific application or category of application to use. generic binary data (or binary data whose true type is unknown) is application/octet-stream. other common examples include application/pdf, application/pkcs8, and application/zip. ...
...esn't fall explicitly into one of the other types; either data that will be executed or interpreted in some way or binary data that requires a specific application or category of application to use. generic binary data (or binary data whose true type is unknown) is application/octet-stream. other common examples include application/pdf, application/pkcs8, and application/zip. audio list at iana audio or music data. ...
...other types; either data that will be executed or interpreted in some way or binary data that requires a specific application or category of application to use. generic binary data (or binary data whose true type is unknown) is application/octet-stream. other common examples include application/pdf, application/pkcs8, and application/zip. audio list at iana audio or music data. examples include audio/mpeg, audio/vorbis. ...
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DynamicsCompressorNode.attack - Web APIs
the attack property of the dynamicscompressornode interface is a k-rate audioparam representing the amount of time, in seconds, required to reduce the gain by 10 db. it defines how quickly the signal is adapted when its volume is increased. the attack property's default value is 0.003 and it can be set between 0 and 1. syntax var audioctx = new audiocontext(); var compressor = audioctx.createdynamicscompressor(); compressor.attack.value = 0; value an audioparam. note: though the audioparam returned is read-only, the value it represents is not. example the below code demonstrates a simple usage of createdynamicscompressor() to add compression to an audio track. for a more complete example, have a look at our basic compressor example (view the source code). // create a mediaelement...
...micscompressor(); compressor.attack.value = 0; value an audioparam. note: though the audioparam returned is read-only, the value it represents is not. example the below code demonstrates a simple usage of createdynamicscompressor() to add compression to an audio track. for a more complete example, have a look at our basic compressor example (view the source code). ...
...ram. note: though the audioparam returned is read-only, the value it represents is not. example the below code demonstrates a simple usage of createdynamicscompressor() to add compression to an audio track. for a more complete example, have a look at our basic compressor example (view the source code). ...
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DynamicsCompressorNode.knee - Web APIs
the knee property of the dynamicscompressornode interface is a k-rate audioparam containing a decibel value representing the range above the threshold where the curve smoothly transitions to the compressed portion. the knee property's default value is 30 and it can be set between 0 and 40. syntax var audioctx = new audiocontext(); var compressor = audioctx.createdynamicscompressor(); compressor.knee.value = 40; value an audioparam. note: though the audioparam returned is read-only, the value it represents is not. example the below code demonstrates a simple usage of createdynamicscompressor() to add compression to an audio track. ...
...amicscompressor(); compressor.knee.value = 40; value an audioparam. note: though the audioparam returned is read-only, the value it represents is not. example the below code demonstrates a simple usage of createdynamicscompressor() to add compression to an audio track. for a more complete example, have a look at our basic compressor example (view the source code). ...
...ram. note: though the audioparam returned is read-only, the value it represents is not. example the below code demonstrates a simple usage of createdynamicscompressor() to add compression to an audio track. for a more complete example, have a look at our basic compressor example (view the source code). ...
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DynamicsCompressorNode.ratio - Web APIs
the ratio property of the dynamicscompressornode interface is a k-rate audioparam representing the amount of change, in db, needed in the input for a 1 db change in the output. the ratio property's default value is 12 and it can be set between 1 and 20. syntax var audioctx = new audiocontext(); var compressor = audioctx.createdynamicscompressor(); compressor.ratio.value = 12; value an audioparam. note: though the audioparam returned is read-only, the value it represents is not. example the below code demonstrates a simple usage of createdynamicscompressor() to add compression to an audio track. for a more complete example, have a look at our basic compressor example (view the source code). ...
...micscompressor(); compressor.ratio.value = 12; value an audioparam. note: though the audioparam returned is read-only, the value it represents is not. example the below code demonstrates a simple usage of createdynamicscompressor() to add compression to an audio track. for a more complete example, have a look at our basic compressor example (view the source code). ...
...ram. note: though the audioparam returned is read-only, the value it represents is not. example the below code demonstrates a simple usage of createdynamicscompressor() to add compression to an audio track. for a more complete example, have a look at our basic compressor example (view the source code). ...
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DynamicsCompressorNode.release - Web APIs
the release property of the dynamicscompressornode interface is a k-rate audioparam representing the amount of time, in seconds, required to increase the gain by 10 db. it defines how quick the signal is adapted when its volume is reduced. the release property's default value is 0.25 and it can be set between 0 and 1. syntax var audioctx = new audiocontext(); var compressor = audioctx.createdynamicscompressor(); compressor.release.value = 0.25; value an audioparam. note: though the audioparam returned is read-only, the value it represents is not. example the below code demonstrates a simple usage of createdynamicscompressor() to add compression to an audio track. for a more complete example, have a look at our basic compressor example (view the source code). // create a mediaelem...
...compressor(); compressor.release.value = 0.25; value an audioparam. note: though the audioparam returned is read-only, the value it represents is not. example the below code demonstrates a simple usage of createdynamicscompressor() to add compression to an audio track. for a more complete example, have a look at our basic compressor example (view the source code). ...
...ram. note: though the audioparam returned is read-only, the value it represents is not. example the below code demonstrates a simple usage of createdynamicscompressor() to add compression to an audio track. for a more complete example, have a look at our basic compressor example (view the source code). ...
...And 8 more matches

DynamicsCompressorNode.threshold - Web APIs
the threshold property of the dynamicscompressornode interface is a k-rate audioparam representing the decibel value above which the compression will start taking effect. the threshold property's default value is -24 and it can be set between -100 and 0. syntax var audioctx = new audiocontext(); var compressor = audioctx.createdynamicscompressor(); compressor.threshold.value = -50; value an audioparam. note: though the audioparam returned is read-only, the value it represents is not. example the below code demonstrates a simple usage of createdynamicscompressor() to add compression to an audio track. for a more complete example, have a look at our basic compressor example (view the source code). ...
...ompressor(); compressor.threshold.value = -50; value an audioparam. note: though the audioparam returned is read-only, the value it represents is not. example the below code demonstrates a simple usage of createdynamicscompressor() to add compression to an audio track. for a more complete example, have a look at our basic compressor example (view the source code). ...
...ram. note: though the audioparam returned is read-only, the value it represents is not. example the below code demonstrates a simple usage of createdynamicscompressor() to add compression to an audio track. for a more complete example, have a look at our basic compressor example (view the source code). ...
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MediaDevices.getUserMedia() - Web APIs
the mediadevices.getusermedia() method prompts the user for permission to use a media input which produces a mediastream with tracks containing the requested types of media. that stream can include, for example, a video track (produced by either a hardware or virtual video source such as a camera, video recording device, screen sharing service, and so forth), an audio track (similarly, produced by a physical or virtual audio source like a microphone, a/d converter, or the like), and possibly other track types. it returns a promise that resolves to a mediastream object. if the user denies permission, or matching media is not available, then the promise is rejected with notallowederror or notfounderror respectively. note: it's possible for the returned promise to neither resolve nor reject, a...
...or more information on this and other security issues related to using getusermedia(). syntax var promise = navigator.mediadevices.getusermedia(constraints); parameters constraints a mediastreamconstraints object specifying the types of media to request, along with any requirements for each type. ...
...ng with any requirements for each type. the constraints parameter is a mediastreamconstraints object with two members: video and audio, describing the media types requested. either or both must be specified. if the browser cannot find all media tracks with the specified types that meet the constraints given, then the returned promise is rejected with notfounderror. the following requests both audio and video without any specific requirements: { audio: true, video: true } if true is specified for a media type, the resulting stream is required to have that type of track in it. if one cannot be included for any reason, the call to getusermedia() will result in an error. while information about a user's cameras and microphones are inaccessible for privacy reasons, an application can reque...
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Media Capabilities API - Web APIs
the media capabilities api allows developers to determine decoding and encoding abilities of the device, exposing information such as whether media is supported and whether playback should be smooth and power efficient, with real time feedback about playback to better enable adaptive streaming, and access to display property information. ...
...allows developers to determine decoding and encoding abilities of the device, exposing information such as whether media is supported and whether playback should be smooth and power efficient, with real time feedback about playback to better enable adaptive streaming, and access to display property information. ...
...ack to better enable adaptive streaming, and access to display property information. examples detect audio file support and expected performance this example defines a audio configuration then checks to see if the user agent supports decoding that media configuration, and whether it will perform well in terms of smoothness and power efficiency. ...
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OscillatorNode - Web APIs
the oscillatornode interface represents a periodic waveform, such as a sine wave. it is an audioscheduledsourcenode audio-processing module that causes a specified frequency of a given wave to be created—in effect, a constant tone. an oscillatornode is created using the baseaudiocontext.createoscillator() method. it always has exactly one output and no inputs. ...
... the oscillatornode interface represents a periodic waveform, such as a sine wave. it is an audioscheduledsourcenode audio-processing module that causes a specified frequency of a given wave to be created—in effect, a constant tone. an oscillatornode is created using the baseaudiocontext.createoscillator() method. it always has exactly one output and no inputs. ...
...outputs 1 channel count mode max channel count 2 (not used in the default count mode) channel interpretation speakers constructor oscillatornode() creates a new instance of an oscillatornode object, optionally providing an object specifying default values for the node's properties. if the default values are acceptable, you can simply call the baseaudiocontext.createoscillator() factory method. properties inherits properties from its parent, audioscheduledsourcenode, and adds the following properties: oscillatornode.frequency an a-rate audioparam representing the frequency of oscillation in hertz (though the audioparam returned is read-only, the value it represents is not). the default value is 440 hz (a standard middle-a note). oscil...
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ScriptProcessorNode.bufferSize - Web APIs
note: as of the august 29 2014 web audio api spec publication, this feature has been marked as deprecated, and is soon to be replaced by audio workers. the buffersize property of the scriptprocessornode interface returns an integer representing both the input and output buffer size, in sample-frames. its value can be a power of 2 value in the range 256–16384. syntax var audioctx = new audiocontext(); var scriptnode = audioctx.createscriptprocessor(4096, 1, 1); console.log(scriptnode.buffersize); value an integer. example the following example shows basic usage of a scriptprocessornode to take a track loaded via audiocontext.decodeaudiodata(), process it, adding a bit of white noise to each audio sample of the input track (buffer) and play it through the audiodestinationnode. for e...
...sor(4096, 1, 1); console.log(scriptnode.buffersize); value an integer. example the following example shows basic usage of a scriptprocessornode to take a track loaded via audiocontext.decodeaudiodata(), process it, adding a bit of white noise to each audio sample of the input track (buffer) and play it through the audiodestinationnode. for each channel and each sample frame, the scriptnode.onaudioprocess function takes the associated audioprocessingevent and uses it to loop through each channel of the input buffer, and each sample in each channel, and add a small amount of white noise, before setting that result to be the output sample in each case. note: for a full working example, see our script-processor-node github repo (also view the source code.) var myscript = document.querysele...
...and each sample in each channel, and add a small amount of white noise, before setting that result to be the output sample in each case. ...
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StereoPannerNode.pan - Web APIs
the pan property of the stereopannernode interface is an a-rate audioparam representing the amount of panning to apply. the value can range between -1 (full left pan) and 1 (full right pan). syntax var audioctx = new audiocontext(); var pannode = audioctx.createstereopanner(); pannode.pan.value = -0.5; returned value an a-rate audioparam containing the panning to apply. note: though the audioparam returned is read-only, the value it represents is not. example in our stereopannernode example (see source code) html we have a simple <audio> element along with a slider <input> to increase and decrease pan value. in the javascript we create a mediaelementaudiosourcenode and a stereopannernode, and connect the two together using the connect() method. we then use an oninput event handler ...
...taudiosourcenode and a stereopannernode, and connect the two together using the connect() method. we then use an oninput event handler to change the value of the stereopannernode.pan parameter and update the pan value display when the slider is moved. moving the slider left and right while the music is playing pans the music across to the left and right speakers of the output, respectively. ...
...stereopannernode, and connect the two together using the connect() method. we then use an oninput event handler to change the value of the stereopannernode.pan parameter and update the pan value display when the slider is moved. moving the slider left and right while the music is playing pans the music across to the left and right speakers of the output, respectively. ...
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Web APIs
a angle_instanced_arrays abortcontroller abortsignal absoluteorientationsensor abstractrange abstractworker accelerometer addresserrors aescbcparams aesctrparam...
... the interfaces (that is, types of objects) that are available. ...
...ces (that is, types of objects) that are available. ...
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Index - MDN Web Docs Glossary: Definitions of Web-related terms
integrity, availability) (also called the cia triad or aic triad) is a model that guides an organization's policies for information security. 47 cms cms, composing, content management system, glossary a cms (content management system) is software that allows users to publish, organize, change, or remove various kinds of content, not only text but also embedded images, video, audio, and interactive code. 48 cors glossary, infrastructure, security cors (cross-origin resource sharing) is a system, consisting of transmitting http headers, that determines whether browsers block frontend javascript code from accessing responses for cross-origin requests. 49 cors-safelisted request header cors, fetch a cors-safelisted request header is on...
...rivate information. the concept of a constructor can be applied to most object-oriented programming languages. essentially, a constructor in javascript is usually declared at the instance of a class. 89 continuous media glossary, media continuous media is data where there is a timing relationship between source and destination. the most common examples of continuous media are audio and motion video. continuous media can be real-time (interactive), where there is a "tight" timing relationship between source and sink, or streaming (playback), where the relationship is less strict. 90 control flow codingscripting, glossary, javascript the control flow is the order in which the computer executes statements in a script. 91 cookie glossary, web...
...assigned names and numbers) is an international nonprofit that maintains the domain name system and the record of ip addresses. 213 ice codingscripting, glossary, networking, protocols, webrtc ice (interactive connectivity establishment) is a framework used by webrtc (among other technologies) for connecting two peers to each other, regardless of network topology (usually for audio and/or video chat). this protocol lets two peers find and establish a connection with one another even though they may both be using network address translator (nat) to share a global ip address with other devices on their respective local networks. 214 ide codingscripting, glossary an integrated development environment (ide) or interactive development environment is a softwa...
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AnalyserNode - Web APIs
an analysernode has exactly one input and one output. the node works even if the output is not connected. ...
... an analysernode has exactly one input and one output. the node works even if the output is not connected. ...
... an analysernode has exactly one input and one output. the node works even if the output is not connected. ...
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DelayNode - Web APIs
the delaynode interface represents a delay-line; an audionode audio-processing module that causes a delay between the arrival of an input data and its propagation to the output. a delaynode always has exactly one input and one output, both with the same amount of channels. when creating a graph that has a cycle, it is mandatory to have at least one delaynode in the cycle, or the nodes taking part in the cycle will be muted. number of inputs 1 number of outputs 1 channel count mode "max" channel count 2 (not used in the default count mode) channel interpretation "speakers" constructor delaynode() creates a new instance of an delaynode object instance. ...
...e cycle, or the nodes taking part in the cycle will be muted. number of inputs 1 number of outputs 1 channel count mode "max" channel count 2 (not used in the default count mode) channel interpretation "speakers" constructor delaynode() creates a new instance of an delaynode object instance. alternatively, you can use the baseaudiocontext.createdelay() factory method. properties inherits properties from its parent, audionode. delaynode.delaytime read only is an a-rate audioparam representing the amount of delay to apply, specified in seconds. methods no specific methods; inherits methods from its parent, audionode. example we have created a simple example that allows you to play three different samples on a con...
... have created a simple example that allows you to play three different samples on a constant loop — see create-delay (you can also view the source code). if you just press the play buttons, the loops will start immediately; if you slide the sliders up to the right, then press the play buttons, a delay will be introduced, so the looping sounds don't start playing for a short amount of time. ...
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MediaDevices.ondevicechange - Web APIs
information about the change included in the event object; to get the updated list of devices, you'll have to use enumeratedevices(). example in this example, we create a function called updatedevicelist(), which is called once when mediadevices.getusermedia() successfully obtains a stream, and then is called any time the device list changes. it displays in the browser window two lists: one of audio devices and one of video devices, with both the device's label (name) and whether it's an input or an output device. because the example provides a handler for the devicechange event, the list is refreshed any time a media device is attached to or removed from the device running the sample. html content <p>click the start button below to begin the demonstration.</p> <div id="startbutton" class...
... because the example provides a handler for the devicechange event, the list is refreshed any time a media device is attached to or removed from the device running the sample. ...
...evicechange event, the list is refreshed any time a media device is attached to or removed from the device running the sample. ...
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Using the Screen Capture API - Web APIs
is in effect, the machine which is sharing screen contents will display some form of indicator so the user is aware that sharing is taking place. note: for privacy and security reasons, screen sharing sources are not enumerable using enumeratedevices(). related to this, the devicechange event is never sent when there are changes to the sources available for getdisplaymedia(). ...
...ote: for privacy and security reasons, screen sharing sources are not enumerable using enumeratedevices(). related to this, the devicechange event is never sent when there are changes to the sources available for getdisplaymedia(). capturing shared audio getdisplaymedia() is most commonly used to capture video of a user's screen (or parts thereof). however, user agents may allow the capture of audio along with the video content. ...
...ources are not enumerable using enumeratedevices(). related to this, the devicechange event is never sent when there are changes to the sources available for getdisplaymedia(). capturing shared audio getdisplaymedia() is most commonly used to capture video of a user's screen (or parts thereof). however, user agents may allow the capture of audio along with the video content. the source of this audio might be the selected window, the entire computer's audio system, or the user's microphone (or a combination of all of the above). before starting a project that will require sharing of audio, be sure to check the browser compatibility in mediadevices.getdisplaymedia() to see if the browsers you wish compaibility with have support for audio in captured screen streams. to request that the screen...
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GainNode - Web APIs
the gainnode interface represents a change in volume. it is an audionode audio-processing module that causes a given gain to be applied to the input data before its propagation to the output. a gainnode always has exactly one input and one output, both with the same number of channels. the gain is a unitless value, changing with time, that is multiplied to each corresponding sample of all input channels. if modified, the new gain is instantly applied, causing unaesthetic 'clicks' in the resulting audio. to prevent this from happening, never change the value directly but use the exponential interpolation methods on the audioparam interface. number of inputs 1 number of outputs 1 channel count mode "max" channel count 2 (not used in the default...
...on methods on the audioparam interface. number of inputs 1 number of outputs 1 channel count mode "max" channel count 2 (not used in the default count mode) channel interpretation "speakers" constructor gainnode() creates a new instance of a gainnode object. you shouldn't manually create a gain node; instead, use the method audiocontext.creategain(). properties inherits properties from its parent, audionode. gainnode.gain read only is an a-rate audioparam representing the amount of gain to apply. you have to set audioparam.value or use the methods of audioparam to change the effect of gain. methods no specific method; inherits methods from its parent, audionode. example the following example shows basic usage...
....creategain(). properties inherits properties from its parent, audionode. gainnode.gain read only is an a-rate audioparam representing the amount of gain to apply. you have to set audioparam.value or use the methods of audioparam to change the effect of gain. methods no specific method; inherits methods from its parent, audionode. example the following example shows basic usage of an audiocontext to create a gainnode, which is then used to mute and unmute the audio when a mute button is clicked by changing the gain property value. ...
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MediaError.message - Web APIs
error.message; value a domstring providing a detailed, specific explanation of what went wrong and possibly how it might be fixed. this is not simply a generic description of the mediaerror.code property's value, but instead goes deeper into the specifics of this particular error and its circumstances. if no specific details are available, this string is empty. example this example creates a <audio> element, establishes an error handler for it, then lets the user click buttons to choose whether to assign a valid audio file or a missing file to the element's src attribute. the error handler simply outputs a message to a box onscreen describing the error, including both the code and the message. only the relevant parts of the code are displayed; you can see the complete source code here. th...
... an error handler for it, then lets the user click buttons to choose whether to assign a valid audio file or a missing file to the element's src attribute. the error handler simply outputs a message to a box onscreen describing the error, including both the code and the message. only the relevant parts of the code are displayed; you can see the complete source code here. the example creates an <audio> element and lets the user assign either a valid music file to it, or a link to a file which doesn't exist. this lets us see the behavior of the error event handler, which is received by an event handler we add to the <audio> element itself. ...
...he relevant parts of the code are displayed; you can see the complete source code here. the example creates an <audio> element and lets the user assign either a valid music file to it, or a link to a file which doesn't exist. this lets us see the behavior of the error event handler, which is received by an event handler we add to the <audio> element itself. ...
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MediaSession.setActionHandler() - Web APIs
ion to the specified time within the media. skipad skips past the currently playing advertisement or commercial. this action may or may not be available, depending on the platform and user agent, or may be disabled due to subscription level or other circumstances. stop halts playback entirely. ...
...ther circumstances. stop halts playback entirely. examples adding action handlers this example implements seek forward and backward actions for an audio player by setting up the seekforward and seekbackward action handlers. let skiptime = 10; // time to skip in seconds navigator.mediasession.setactionhandler('seekforward', evt => { // user clicked "seek forward" media notification icon. audio.currenttime = math.min(audio.currenttime + skiptime, audio.duration); }); navigator.mediasession.setactionhandler('seekbackward', evt => { // user clicked "seek backward" media notification icon. ...
...ndlers. let skiptime = 10; // time to skip in seconds navigator.mediasession.setactionhandler('seekforward', evt => { // user clicked "seek forward" media notification icon. audio.currenttime = math.min(audio.currenttime + skiptime, audio.duration); }); navigator.mediasession.setactionhandler('seekbackward', evt => { // user clicked "seek backward" media notification icon. ...
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Media Session action types - Web APIs
ircumstances. stop halts playback entirely. description a media session action may be generated by any media session action source; these sources include anything from ui widgets within the browser itself to media control keys on the user's keyboard to buttons on the user's headset or earbuds. ...
...eyboard to buttons on the user's headset or earbuds. examples adding action handlers this example implements seek forward and backward actions for an audio player by setting up the seekforward and seekbackward action handlers. let skiptime = 10; // time to skip in seconds navigator.mediasession.setactionhandler('seekforward', evt => { // user clicked "seek forward" media notification icon. audio.currenttime = math.min(audio.currenttime + skiptime, audio.duration); }); navigator.mediasession.setactionhandler('seekbackward', evt => { // user clicked "seek backward" media notification icon. ...
...ndlers. let skiptime = 10; // time to skip in seconds navigator.mediasession.setactionhandler('seekforward', evt => { // user clicked "seek forward" media notification icon. audio.currenttime = math.min(audio.currenttime + skiptime, audio.duration); }); navigator.mediasession.setactionhandler('seekbackward', evt => { // user clicked "seek backward" media notification icon. ...
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MediaSessionActionDetails - Web APIs
ircumstances. stop halts playback entirely. description a media session action may be generated by any media session action source; these sources include anything from ui widgets within the browser itself to media control keys on the user's keyboard to buttons on the user's headset or earbuds. ...
...eyboard to buttons on the user's headset or earbuds. examples adding action handlers this example implements seek forward and backward actions for an audio player by setting up the seekforward and seekbackward action handlers. let skiptime = 10; // time to skip in seconds navigator.mediasession.setactionhandler('seekforward', evt => { // user clicked "seek forward" media notification icon. audio.currenttime = math.min(audio.currenttime + skiptime, audio.duration); }); navigator.mediasession.setactionhandler('seekbackward', evt => { // user clicked "seek backward" media notification icon. ...
...ndlers. let skiptime = 10; // time to skip in seconds navigator.mediasession.setactionhandler('seekforward', evt => { // user clicked "seek forward" media notification icon. audio.currenttime = math.min(audio.currenttime + skiptime, audio.duration); }); navigator.mediasession.setactionhandler('seekbackward', evt => { // user clicked "seek backward" media notification icon. ...
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Example and tutorial: Simple synth keyboard - Web APIs
this article presents the code and working demo of a video keyboard you can play using the mouse. the keyboard allows you to switch among the standard waveforms as well as one custom waveform, and you can control the master gain using a volume slider beneath the keyboard. this example makes use of the following web api interfaces: audiocontext, oscillatornode, periodicwave, and gainnode. because oscillatornode is based on audioscheduledsourcenode, this is to some extent an example for that as well. the video keyboard html there are three primary components to the display for our virtual keyboard. the first is the musical keyboard itself. we draw this in a pair of nested <div> elements so that we can make the keyboard horizontally scrollable if all the keys don't fit on the screen, without ...
... left: 0; display: table-cell; vertical-align: middle; } .left span, .left input { vertical-align: middle; } .right { width: 50%; position: absolute; right: 0; display: table-cell; vertical-align: middle; } .right span { vertical-align: middle; } .right input { vertical-align: baseline; } javascript the javascript code begins by initializing a number of variables. let audiocontext = new (window.audiocontext || window.webkitaudiocontext)(); let osclist = []; let mastergainnode = null; audiocontext is set to reference the global audiocontext object (or webkitaudiocontext if necessary). osclist is set up to be ready to contain a list of all currently-playing oscillators. it starts off empty, since there are none playing yet. mastergainnode is set to null; during ...
...r("select[name='waveform']"); let volumecontrol = document.queryselector("input[name='volume']"); references to elements we'll need access to are obtained: keyboard is the container element into which the keys will be placed. wavepicker is the <select> element used to choose the waveform to use for the notes. volumecontrol is the <input> element (of type "range") used to control the master audio volume. let notefreq = null; let customwaveform = null; let sineterms = null; let cosineterms = null; finally, global variables that will be used when constructing waveforms are created: notefreq will be an array of arrays; each array represents one octave, each of which contains one entry for each note in that octave. the value for each is the frequency, in hertz, of the note's tone. cus...
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Event reference
user starts dragging an element or text selection. dragleave a dragged element or text selection leaves a valid drop target. dragover an element or text selection is being dragged over a valid drop target (fired continuously every 350ms). drop an element is dropped on a valid drop target. media events event name fired when audioprocess the input buffer of a scriptprocessornode is ready to be processed. canplay the browser can play the media, but estimates that not enough data has been loaded to play the media up to its end without having to stop for further buffering of content. canplaythrough the browser estimates it can play the media up to its end without stopping for content buffering. ...
...ssornode is ready to be processed. canplay the browser can play the media, but estimates that not enough data has been loaded to play the media up to its end without having to stop for further buffering of content. canplaythrough the browser estimates it can play the media up to its end without stopping for content buffering. complete the rendering of an offlineaudiocontext is terminated. durationchange the duration attribute has been updated. emptied the media has become empty; for example, this event is sent if the media has already been loaded (or partially loaded), and the load() method is called to reload it. ended playback has stopped because the end of the media was reached. loadeddata the first frame of ...
...nd animationevent css animations a css animation has completed. animationiteration animationevent css animations a css animation is repeated. animationstart animationevent css animations a css animation has started. appinstalled event web app manifest a web application is successfully installed as a progressive web app. ...
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Web media technologies
over the years, the web's ability to present, create, and manage audio, video, and other media has grown at an increasing pace. today, there are a large number of apis available, as well as html elements, dom interfaces, and other features that make it possible to not only perform these tasks, but use media in tandem with other technologies to do truly remarkable things. this article lists the various apis with links to documentation you may find helpful in mastering them. references html these articles cover html features for media developers. <audio> the <audio> element is used to play audio in a web context. these can be used invisibly as a destination for more complex media, or with visible controls for user-controlled playback of audio files. accessible from javascript as htmlaud...
...o content in a web context. it can be used to simply present video files, or as a destination for streamed video content. <video> can also be used as a way to link media apis with other html and dom technologies, including <canvas> (for frame grabbing and manipulation), for example. accessible from javascript as htmlvideoelement objects. <track> the html <track> element can be placed within an <audio> or <video> element to provide a reference to a webvtt format subtitle or caption track to be used when playing the media. accessible from javascript as htmltrackelement objects. <source> the html <source> element is used within an <audio> or <video> element to specify source media to present. multiple sources can be used to provide the media in different formats, sizes, or resolutions. accessi...
...ts you determine the encoding and decoding capabilities of the device your app or site is running on. this lets you make real-time decisions about what formats to use and when. media capture and streams api a reference for the api which makes it possible to stream, record, and manipulate media both locally and across a network. this includes using local cameras and microphones to capture video, audio, and still images. mediastream recording api the mediastream recording api lets you capture media streams to process or filter the data or record it to disk. web audio api the web audio api lets you generate, filter, and manipulate sound data both in real-time and on pre-recorded material, then send that audio to a destination such as an <audio> element, a media stream, or to disk. webrtc w...
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Index - Archive of obsolete content
in the past, this had to be simulated by treating the raw data as a string and using the charcodeat() method to read the bytes from the data buffer. 213 tabbox ...
...tches obsolete adobe and mozilla share a copy of nanojit. this means that landing nanojit patches is a bit complicated, but experience has taught us that this is much better than having separate versions of nanojit. 488 io guide/directory keys "directory keys" are the keys for nsidirectoryservice, see description. the actual keys are: 489 introducing the audio api extension deprecated the audio data api extension extends the html5 specification of the <audio> and <video> media elements by exposing audio metadata and raw audio data. this enables users to visualize audio data, to process this audio data and to create new audio data. 490 isp data administration, enterprise this page will explain how to create a .rdf or .xml ...
...ipt template add-ons, plugins, xpinstall no summary! 714 xpinstall api reference xpinstall, xpinstall_api_reference see complete list 715 examples trigger scripts and install scripts describes typical kinds of script that use the xpinstall api. 716 file.macalias in this example, a mac alias is created for software (applecd audio player) that is installed locally. 717 file.windowsshortcut in this example, the windowsshortcut method is used to add a shortcut in the program directory ("program" is a keyword for the directory in which the program itself is installed, for example, c:\program files\netscape\netscape 6\" on windows) to windows software (misc.exe) that is installed in the "windows" directo...
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Creating a Web based tone generator - Archive of obsolete content
this example creates a simple tone generator, and plays the resulting tone. the function requestsounddata() produces the samples to be played. this function is called at a certain interval through the setinterval() function. the function mozwriteaudio() is called to write those samples produced in the audio stream. in order to always have samples to play, a buffer of 500 ms is created. the function mozcurrentsampleoffset() is used to know the position of the samples being played so that we can fill a 500 ms buffer of audio samples. ...
...gh the setinterval() function. the function mozwriteaudio() is called to write those samples produced in the audio stream. in order to always have samples to play, a buffer of 500 ms is created. the function mozcurrentsampleoffset() is used to know the position of the samples being played so that we can fill a 500 ms buffer of audio samples. <!doctype html> <html> <head> <title>javascript audio write example</title> </head> <body> <input type="text" size="4" id="freq" value="440"><label for="hz">hz</label> <button onclick="start()">play</button> <button onclick="stop()">stop</button> <script type="text/javascript"> function audiodatadestination(samplerate, readfn) { // initialize the audio output. ...
...<html> <head> <title>javascript audio write example</title> </head> <body> <input type="text" size="4" id="freq" value="440"><label for="hz">hz</label> <button onclick="start()">play</button> <button onclick="stop()">stop</button> <script type="text/javascript"> function audiodatadestination(samplerate, readfn) { // initialize the audio output. var audio = new audio(); audio.mozsetup(1, samplerate); var currentwriteposition = 0; var prebuffersize = samplerate / 2; // buffer 500ms var tail = null; // the function called with regular interval to populate // the audio output buffer. setinterval(function() { var written; // check if some data was not written in previous att...
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HTML attribute reference - HTML: Hypertext Markup Language
input is automatically capitalized when entered by user autocomplete <form>, <input>, <select>, <textarea> indicates whether controls in this form can by default have their values automatically completed by the browser. autofocus <button>, <input>, <keygen>, <select>, <textarea> the element should be automatically focused after the page loaded. autoplay <audio>, <video> the audio or video should play as soon as possible. background <body>, <table>, <td>, <th> specifies the url of an image file. note: although browsers and email clients may still support this attribute, it is obsolete. use css background-image instead. bgcolor <body>, <col>, <colgroup>, <marquee>, <table>, <tbody>, <tfoot>, <td>, <th>, <tr> ...
...group>, <marquee>, <table>, <tbody>, <tfoot>, <td>, <th>, <tr> background color of the element. note: this is a legacy attribute. please use the css background-color property instead. border <img>, <object>, <table> the border width. note: this is a legacy attribute. please use the css border property instead. buffered <audio>, <video> contains the time range of already buffered media. capture <input> from the html media capturethe definition of 'media capture' in that specification.spec, specifies a new file can be captured. challenge <keygen> a challenge string that is submitted along with the public key. charset <meta>, <script> declares the character encoding of the ...
... defines the number of columns a cell should span. content <meta> a value associated with http-equiv or name depending on the context. contenteditable global attribute indicates whether the element's content is editable. contextmenu global attribute defines the id of a <menu> element which will serve as the element's context menu. ...
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HTML documentation index - HTML: Hypertext Markup Language
eference the itemprop global attribute is used to add properties to an item. every html element can have an itemprop attribute specified, and an itemprop consists of a name-value pair. each name-value pair is called a property, and a group of one or more properties forms an item. property values are either a string or a url and can be associated with a very wide range of elements including <audio>, <embed>, <iframe>, <img>, <link>, <object>, <source> , <track>, and <video>. 23 itemref attribute, global attribute, html, html microdata, microdata, reference properties that are not descendants of an element with the itemscope attribute can be associated with an item using the global attribute itemref. 24 itemscope attribute, global attribute, html, html mi...
...idation, html the capture attribute specifies that, optionally, a new file should be captured, and which device should be used to capture that new media of a type defined by the accept attribute. values include user and environment. 39 html attribute: crossorigin advanced, attribute, cors, html, needscontent, reference, security the crossorigin attribute, valid on the <audio>, <img>, <link>, <script>, and <video> elements, provides support for cors, defining how the element handles crossorigin requests, thereby enabling the configuration of the cors requests for the element's fetched data. depending on the element, the attribute can be a cors settings attribute. 40 html attribute: max attribute, attributes, constraint validation, html, reference ...
...is intended to be independently distributable or reusable (e.g., in syndication). 62 <aside>: the aside element element, html, html sections, html5, html:flow content, html:palpable content, html:sectioning content, reference, web the html <aside> element represents a portion of a document whose content is only indirectly related to the document's main content. ...
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Using Promises - JavaScript
ise is an object representing the eventual completion or failure of an asynchronous operation. since most people are consumers of already-created promises, this guide will explain consumption of returned promises before explaining how to create them. essentially, a promise is a returned object to which you attach callbacks, instead of passing callbacks into a function. ...
...-created promises, this guide will explain consumption of returned promises before explaining how to create them. essentially, a promise is a returned object to which you attach callbacks, instead of passing callbacks into a function. imagine a function, createaudiofileasync(), which asynchronously generates a sound file given a configuration record and two callback functions, one called if the audio file is successfully created, and the other called if an error occurs. ...
...m. essentially, a promise is a returned object to which you attach callbacks, instead of passing callbacks into a function. imagine a function, createaudiofileasync(), which asynchronously generates a sound file given a configuration record and two callback functions, one called if the audio file is successfully created, and the other called if an error occurs. ...
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2D maze game with device orientation - Game development
forward: the starting point is the index.html file where we initialize the framework and set up an <canvas> to render the game on. you can open the index file in your favourite browser to launch the game and try it. there are also three folders in the directory: img: all the images that we will use in the game. src: the javascript files with all the source code of the game defined inside. audio: the sound files used in the game. setting up the canvas we will be rendering our game on canvas, but we won't do it manually — this will be taken care of by the framework. let’s set it up: our starting point is the index.html file with the following content. you can create this yourself if you want to follow along: <!doctype html> <html> <head> <meta charset="utf-8" /> <title>cyb...
...e what the states are actually. managing game states the states in phaser are separate parts of the game logic; in our case we’re loading them from independent javascript files for better maintainability. the basic states used in this game are: boot, preloader, mainmenu, howto and game. boot will take care of initializing a few settings, preloader will load all of the assets like graphics and audio, mainmenu is the menu with the start button, howto shows the "how to play" instructions and the game state lets you actually play the game. let's quickly go though the content of those states. boot.js the boot state is the first one in the game. var ball = { _width: 320, _height: 480 }; ball.boot = function(game) {}; ball.boot.prototype = { preload: function() { this.load.i...
...((ball._width-297)*0.5, (ball._height-145)*0.5, 'preloaderbg'); this.preloadbar = this.add.sprite((ball._width-158)*0.5, (ball._height-50)*0.5, 'preloaderbar'); this.load.setpreloadsprite(this.preloadbar); this.load.image('ball', 'img/ball.png'); // ... this.load.spritesheet('button-start', 'img/button-start.png', 146, 51); // ... this.load.audio('audio-bounce', ['audio/bounce.ogg', 'audio/bounce.mp3', 'audio/bounce.m4a']); }, create: function() { this.game.state.start('mainmenu'); } }; there are single images, spritesheets and audio files loaded by the framework. in this state the preloadbar is showing the progress on the screen. that progress of the loaded assets is visualized by the framework with the use of one im...
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Index - Learn web development
ponsibly so that all readers might use your creations on the web. to help you achieve this, this module will cover general best practices (which are demonstrated throughout the html, css, and javascript topics), cross browser testing, and some tips on enforcing accessibility from the start. we'll cover accessibility in special detail. 3 accessible multimedia accessibility, article, audio, beginner, codingscripting, html, images, javascript, learn, multimedia, video, captions, subtitles, text tracks this chapter has provided a summary of accessibility concerns for multimedia content, along with some practical solutions. 4 assessment: accessibility troubleshooting accessibility, assessment, beginner, css, codingscripting, html, javascript, learn, wai-aria ...
... importantly, you should understand what the document object model is, and how to manipulate it to create useful functionality. 62 third-party apis 3rd party, api, beginner, codingscripting, google maps, learn, nytimes, third party, youtube this article has given you a useful introduction to using third-party apis to add functionality to your websites. 63 video and audio apis api, article, audio, beginner, codingscripting, guide, javascript, learn, video i think we've taught you enough in this article. the htmlmediaelement api makes a wealth of functionality available for creating simple video and audio players, and that's only the tip of the iceberg. see the "see also" section below for links to more complex and interesting functionality. 64 ...
... metadata that marks the end of our quickfire tour of the html head — there's a lot more you can do in here, but an exhaustive tour would be boring and confusing at this stage, and we just wanted to give you an idea of the most common things you'll find in there for now! in the next article we'll be looking at html text fundamentals. 243 multimedia and embedding assessment, audio, beginner, codingscripting, embedding, flash, guide, html, images, landing, learn, multimedia, svg, vector graphics, video, web, iframes, imagemaps, img, responsive we've looked at a lot of text so far in this course, but the web would be really boring only using text. let's start looking at how to make the web come alive with more interesting content! this module explores how to use html t...
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ChannelMergerNode - Web APIs
to a single output. each input is used to fill a channel of the output. this is useful for accessing each channels separately, e.g. for performing channel mixing where gain must be separately controlled on each channel. if channelmergernode has one single output, but as many inputs as there are channels to merge; the number of inputs is defined as a parameter of its constructor and the call to audiocontext.createchannelmerger. in the case that no value is given, it will default to 6. using a channelmergernode, it is possible to create outputs with more channels than the rendering hardware is able to process. in that case, when the signal is sent to the audiocontext.listener object, supernumerary channels will be ignored. number of inputs variable; default to 6. number ...
... number of inputs variable; default to 6. number of outputs 1 channel count mode "max" channel count 2 (not used in the default count mode) channel interpretation "speakers" constructor channelmergernode() creates a new channelmergernode object instance. ...
...utputs 1 channel count mode "max" channel count 2 (not used in the default count mode) channel interpretation "speakers" constructor channelmergernode() creates a new channelmergernode object instance. properties no specific property; inherits properties from its parent, audionode. methods no specific method; inherits methods from its parent, audionode. ...
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ConstantSourceNode.offset - Web APIs
the default value is 1.0. ...
...the default value is 1.0. ...
...the default value is 1.0. to access the offset parameter's current value, access the parameter's value property, as shown in the syntax box above. ...
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ConstantSourceNode - Web APIs
a constantsourcenode has no inputs and exactly one monaural (one-channel) output. the output's value is always the same as the value of the offset parameter. ...
... a constantsourcenode has no inputs and exactly one monaural (one-channel) output. the output's value is always the same as the value of the offset parameter. ...
... a constantsourcenode has no inputs and exactly one monaural (one-channel) output. the output's value is always the same as the value of the offset parameter. ...
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ConvolverNode.normalize - Web APIs
buffer attribute is set, or not. its default value is true in order to achieve a more uniform output level from the convolver, when loaded with diverse impulse responses. if normalize is set to false, then the convolution will be rendered with no pre-processing/scaling of the impulse response. changes to this value do not take effect until the next time the buffer attribute is set. ...
...te is set, or not. its default value is true in order to achieve a more uniform output level from the convolver, when loaded with diverse impulse responses. if normalize is set to false, then the convolution will be rendered with no pre-processing/scaling of the impulse response. changes to this value do not take effect until the next time the buffer attribute is set. ...
...value is true in order to achieve a more uniform output level from the convolver, when loaded with diverse impulse responses. if normalize is set to false, then the convolution will be rendered with no pre-processing/scaling of the impulse response. changes to this value do not take effect until the next time the buffer attribute is set. ...
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GainNode.gain - Web APIs
the gain property of the gainnode interface is an a-rate audioparam representing the amount of gain to apply. syntax var audioctx = new audiocontext(); var gainnode = audioctx.creategain(); gainnode.gain.value = 0.5; value an audioparam. note: though the audioparam returned is read-only, the value it represents is not. example the following example shows basic usage of an audiocontext to create a gainnode, which is then used to mute and unmute the audio when a mute button is clicked by changing the gain property value. ...
... example the following example shows basic usage of an audiocontext to create a gainnode, which is then used to mute and unmute the audio when a mute button is clicked by changing the gain property value. ...
...g example shows basic usage of an audiocontext to create a gainnode, which is then used to mute and unmute the audio when a mute button is clicked by changing the gain property value. ...
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Using the Media Capabilities API - Web APIs
taking video as an example, to obtain information about video decoding abilities, you create a video decoding configuration which you pass as a parameter to mediacapabilities.decodinginfo() method. this returns a promise that fulfills with information about the media capabilities as to whether the video can be decoded, and whether decoding will be smooth and power efficient. you can also test audio decoding as well as video and audio encoding. creating a video decoding configuration the mediacapabilities.decodinginfo() method takes as a parameter a media decoding configuration. there are very specific ways to go about creating the configuration defined by the mediadecodingconfiguration dictionary. in our example, we are testing the decoding capabilities of a video configuration. the conf...
...itrate is the number of bits used to encode one second of video. the framerate is the number of frames which are played per second of time when playing the video. ...
...ed to encode one second of video. the framerate is the number of frames which are played per second of time when playing the video. ...
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RTCPeerConnection.createOffer() - Web APIs
y is used to customize the offer created by this method. icerestart optional to restart ice on an active connection, set this to true. this will cause the returned offer to have different credentials than those already in place. if you then apply the returned offer, ice will restart. specify false to keep the same credentials and therefore not restart ice. the default is false. ...
...ill cause the returned offer to have different credentials than those already in place. if you then apply the returned offer, ice will restart. specify false to keep the same credentials and therefore not restart ice. the default is false. offertoreceiveaudio optional (legacy) a legacy boolean option which used to control whether or not to offer to the remote peer the opportunity to try to send audio. if this value is false, the remote peer will not be offered to send audio data, even if the local side will be sending audio data. if this value is true, the remote peer will be offered to send audio data, even if the local side will not be sending audio data. the default behavior is to offer to receive audio only if the local side is sending audio, not otherwise. to emulate this behavior in mo...
... note: you shouldn't use this legacy property. ...
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Worklet.addModule() - Web APIs
dentials. return value a promise that resolves once the module from the given url has been added. the promise doesn't return any value. exceptions if addmodule() fails, it rejects the promise, delivering one of the following errors to the rejection handler. aborterror the specified script is invalid or could not be loaded. syntaxerror the specified moduleurl is invalid. ...
... a promise that resolves once the module from the given url has been added. the promise doesn't return any value. exceptions if addmodule() fails, it rejects the promise, delivering one of the following errors to the rejection handler. aborterror the specified script is invalid or could not be loaded. syntaxerror the specified moduleurl is invalid. ...
...t resolves once the module from the given url has been added. the promise doesn't return any value. exceptions if addmodule() fails, it rejects the promise, delivering one of the following errors to the rejection handler. aborterror the specified script is invalid or could not be loaded. syntaxerror the specified moduleurl is invalid. ...
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ARIA: button role - Accessibility
-pressed is not used, or is set to the "undefined" state, the button is not a toggle button. if aria-pressed="false" is used the button is a toggle button that is currently not pressed. if aria-pressed="true" is used the button is a toggle button that is currently pressed. if aria-pressed="mixed" is used, the button is considered to be partially pressed. as an example, the mute button on an audio player labeled "mute" could indicate that sound is muted by setting the aria-pressed state true. the label of a toggle button should not change when its state changes. in our example the label remains "mute" with a screen reader reading "mute toggle button pressed" or "mute toggle button not pressed" depending on the value of aria-pressed. if the design were to call for the button label to change...
...on performs. for example, if clicking the button opens a dialog, the focus should move to the dialog. if the button closes a dialog, focus should returns to the button that opened the dialog unless the function performed in the dialog context logically leads to a different element. if the button alters the current context, then focus typically remains on the button, such as muting and unmuting an audio file. required javascript features required event handlers buttons can be operated by mouse, touch, and keyboard users. for native html <button> elements, the button's onclick event fires for mouse clicks and when the user presses space or enter while the button has focus. but if another tag is used to create a button, the onclick event only fires when clicked by the mouse cursor, even if role...
...; list.appendchild(listitem); } } toggle button example in this snippet a <span> element is converted to a toggle button using the button role and the aria-pressed attribute. when the button is activated, the aria-pressed value switches states; changing from true to false and back again. ...
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Common MIME types - HTTP
n file type should use this type. browsers pay a particular care when manipulating these files, attempting to safeguard the user to prevent dangerous behaviors. iana is the official registry of mime media types and maintains a list of all the official mime types. ...
...pe should use this type. browsers pay a particular care when manipulating these files, attempting to safeguard the user to prevent dangerous behaviors. iana is the official registry of mime media types and maintains a list of all the official mime types. ...
...egistry of mime media types and maintains a list of all the official mime types. ...
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Web video codec guide - Web media technologies
000 bytes (~249 mb). a minute of hd video would need 14.93 gb of storage. a fairly typical 30 minute video conference would need about 447.9 gb of storage, and a 2-hour movie would take almost 1.79 tb (that is, 1790 gb). not only is the required storage space enormous, but the network bandwidth needed to transmit an uncompressed video like that would be enormous, at 249 mb/sec—not including audio and overhead. this is where video codecs come in. just as audio codecs do for the sound data, video codecs compress the video data and encode it into a format that can later be decoded and played back or edited. most video codecs are lossy, in that the decoded video does not precisely match the source. some details may be lost; the amount of loss depends on the codec and how it's configured, but...
...ered by patents. mp4v-es the mpeg-4 video elemental stream (mp4v-es) format is part of the mpeg-4 part 2 visual standard. while in general, mpeg-4 part 2 video is not used by anyone because of its lack of compelling value related to other codecs, mp4v-es does have some usage on mobile. mp4v is essentially h.263 encoding in an mpeg-4 container. its primary purpose is to be used to stream mpeg-4 audio and video over an rtp session. however, mp4v-es is also used to transmit mpeg-4 audio and video over a mobile connection using 3gp. you almost certainly don't want to use this format, since it isn't supported in a meaningful way by any major browsers, and is quite obsolete. files of this type should have the extension .mp4v, but sometimes are inaccurately labeled .mp4. supported bit ra...
...ground able to be seen through the video to a degree specified by each pixel's alpha component. there is good browser support for vp8 in html content, especially within webm files. this makes vp8 a good candidate for your content, although vp9 is an even better choice if available to you. web browsers are required to support vp8 for webrtc, but not all browsers that do so also support it in html audio and video elements. supported bit rates arbitrary; no maximum unless level-based limitations are enforced supported frame rates arbitrary compression lossy dct-based algorithm supported frame sizes up to 16,384 x 16,384 pixels supported color modes y'cbcr with 4:2:0 chroma subsampling at 8 bits per component hdr support no ...
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Multimedia: video - Learn web development
serving the smallest video file size based on each browsers file type support. why optimize your multimedia? for the average website, 25% of bandwidth comes from video. optimizing video has the potential for very large bandwidth savings that translate into better website performance. optimizing video delivery it's best to compress all video, optimize <source> order, set autoplay, remove audio from muted video, optimize video preload, and consider streaming the video. the sections below describe each of these optimization techniques. compress all videos most video compression work compares adjacent frames within a video, with the intent of removing detail that is identical in both frames. compress the video and export to multiple video formats, including webm, mpeg-4/h.264, and ogg/t...
...al attributes to the video tag: autoplay, muted, and playsinline. <video autoplay="" loop="" muted="true" playsinline="" src="backgroundvideo.mp4"> while the loop and autoplay make sense for a looping and autoplaying video, the muted attribute is required for autoplay in mobile browsers. playsinline is required for mobile safari, allowing videos to play without forcing fullscreen mode. remove audio from muted hero videos for hero-video or other video without audio, removing audio is smart. <video autoplay="" loop="" muted="true" playsinline="" id="hero-video"> <source src="banner_video.webm" type='video/webm; codecs="vp8, vorbis"'> <source src="web_banner.mp4" type="video/mp4"> </video> this hero-video code (above) is common to conference websites and corporate home pages. ...
..."" loop="" muted="true" playsinline="" id="hero-video"> <source src="banner_video.webm" type='video/webm; codecs="vp8, vorbis"'> <source src="web_banner.mp4" type="video/mp4"> </video> this hero-video code (above) is common to conference websites and corporate home pages. it includes a video that is auto-playing, looping, and muted. there are no controls, so there is no way to hear audio. ...
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Embedding API for Accessibility
tring); setcharpref("browser.selection.border", cssborderstylestring); /* is a selection border hard to implement? ...
...tylestring); /* is a selection border hard to implement? ...
...ag requirement */ setboolpref("browser.selection.use_system_colors", usesystemcolors); no content waiting alerts setcharpref("alert.audio.mail_waiting", pathtosoundfile); setcharpref("alert.audio.background_image_waiting", pathtosoundfile); setcharpref("alert.audio.popup_waiting", pathtosoundfile); setcharpref("alert.audio.applet_waiting", pathtosoundfile); setcharpref("alert.audio.script_waiting", pathtosoundfile); setcharpref("alert.audio.redirect_waiting", pathtosoundfile); setcharpref("alert.audio.refresh_waiting", pathtosoundfile); setchar...
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ChannelSplitterNode - Web APIs
in the case that no value is given, it will default to 6. if there are fewer channels in the input than there are outputs, supernumerary outputs are silent. ...
...in the case that no value is given, it will default to 6. if there are fewer channels in the input than there are outputs, supernumerary outputs are silent. number of inputs 1 number of outputs variable; default to 6. ...
... of the spec use "max". channel count fixed to the number of outputs. older implementations, as per earlier versions of the spec use 2 (not used in the default count mode). channel interpretation "discrete" constructor channelsplitternode() creates a new channelsplitternode object instance. ...
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IIRFilterNode() - Web APIs
the filter needs these values to work and with the vast range of filters available, there is no default. ...
...the filter needs these values to work and with the vast range of filters available, there is no default. ...
...the filter needs these values to work and with the vast range of filters available, there is no default. ...
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MediaRecorder() - Web APIs
record using a specific media container (file type), and, further, can specify the exact codec and codec configuration(s) to use by specifying the codecs parameter. syntax var mediarecorder = new mediarecorder(stream[, options]); parameters stream the mediastream that will be recorded. ...
.... this source media can come from a stream created using navigator.mediadevices.getusermedia() or from an <audio>, <video> or <canvas> element. options optional a dictionary object that can contain the following properties: mimetype: a mime type specifying the format for the resulting media; you may simply specify the container format (the browser will select its preferred codecs for audio and/or video), or you may use the codecs parameter and/or the profiles parameter to provide detailed information about which codecs to use and how to configure them. applications can check in advance if a mimetype is supported by the user agent by calling mediarecorder.istypesupported(). audiobitspersecond: the chosen bitrate for the audio component of the media. videobitspersecond: the chose...
... detailed information about which codecs to use and how to configure them. applications can check in advance if a mimetype is supported by the user agent by calling mediarecorder.istypesupported(). audiobitspersecond: the chosen bitrate for the audio component of the media. videobitspersecond: the chosen bitrate for the video component of the media. bitspersecond: the chosen bitrate for the audio and video components of the media. this can be specified instead of the above two properties. ...
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MediaTrackSettings - Web APIs
however, be used for initially selecting media when calling mediadevices.getusermedia(). groupid a domstring indicating the current value of the groupid property. the group id is a browsing session-unique string identifying the source group of the track. two devices (as identified by the deviceid) are considered part of the same group if they are from the same physical device. for instance, the audio input and output devices for the speaker and microphone built into a phone would share the same group id, since they're part of the same physical device. the microphone on a headset would have a different id, though. this value is specific to the source of the track's data and is not usable for setting constraints; it can, however, be used for initially selecting media when calling mediadevices.g...
...he speaker and microphone built into a phone would share the same group id, since they're part of the same physical device. the microphone on a headset would have a different id, though. this value is specific to the source of the track's data and is not usable for setting constraints; it can, however, be used for initially selecting media when calling mediadevices.getusermedia(). ...
...this is 1 for mono, 2 for stereo, and so forth. ...
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OscillatorNode.detune - Web APIs
the detune property of the oscillatornode interface is an a-rate audioparam representing detuning of oscillation in cents. syntax var oscillator = audioctx.createoscillator(); oscillator.detune.setvalueattime(100, audioctx.currenttime); // value in cents note: though the audioparam returned is read-only, the value it represents is not. value an a-rate audioparam. example the following example shows basic usage of an audiocontext to create an oscillator node. for applied examples/information, check out our violent theremin demo (see app.js for relevant code). ...
...detune.setvalueattime(100, audioctx.currenttime); // value in cents note: though the audioparam returned is read-only, the value it represents is not. value an a-rate audioparam. example the following example shows basic usage of an audiocontext to create an oscillator node. for applied examples/information, check out our violent theremin demo (see app.js for relevant code). ...
...100, audioctx.currenttime); // value in cents note: though the audioparam returned is read-only, the value it represents is not. value an a-rate audioparam. example the following example shows basic usage of an audiocontext to create an oscillator node. for applied examples/information, check out our violent theremin demo (see app.js for relevant code). ...
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PannerNode.coneOuterGain - Web APIs
the default is 0, and its value can be in the range 0–1. ...
...the default is 0, and its value can be in the range 0–1. ...
...the default is 0, and its value can be in the range 0–1. ...
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PannerNode.orientationX - Web APIs
the orientationx property of the pannernode interface indicates the x (horizontal) component of the direction in which the audio source is facing, in a 3d cartesian coordinate space. the complete vector is defined by the position of the audio source, given as (positionx, positiony, positionz), and the orientation of the audio source (that is, the direction in which it's facing), given as (orientationx, orientationy, orientationz). depending on the directionality of the sound (as specified using the attributes coneinnerangle, coneouterangle, and coneoutergain), the orientation of the sound may alter the perceived volume of the sound as it's being played. ...
...en as (orientationx, orientationy, orientationz). depending on the directionality of the sound (as specified using the attributes coneinnerangle, coneouterangle, and coneoutergain), the orientation of the sound may alter the perceived volume of the sound as it's being played. if the sound is pointing toward the listener, it will be louder than if the sound is pointed away from the listener. ...
...ngle, coneouterangle, and coneoutergain), the orientation of the sound may alter the perceived volume of the sound as it's being played. if the sound is pointing toward the listener, it will be louder than if the sound is pointed away from the listener. the audioparam contained by this property is read only; however, you can still change the value of the parameter by assigning a new value to its audioparam.value property. syntax var orientationx = pannernode.orientationx; pannernode.orientationx.value = neworientationx; value an audioparam whose value is the x component of the direction in which the audio source is facing, in 3d cartesian coordinate space. example in this example, we'll demonstrate how changing the orientation parameters of a pannernode in combination with coneinneran...
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PannerNode.positionZ - Web APIs
depending on the directionality of the sound (as specified using the attributes coneinnerangle, coneouterangle, and codeoutergain), the orientation of the sound may alter the perceived volume of the sound as it's being played. ...
... depending on the directionality of the sound (as specified using the attributes coneinnerangle, coneouterangle, and codeoutergain), the orientation of the sound may alter the perceived volume of the sound as it's being played. ...
... depending on the directionality of the sound (as specified using the attributes coneinnerangle, coneouterangle, and codeoutergain), the orientation of the sound may alter the perceived volume of the sound as it's being played. ...
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PannerNode.refDistance - Web APIs
if the value is set to less than 0, a rangeerror is thrown. exceptions rangeerror the property has been given a value that is outside the accepted range. ...
...if the value is set to less than 0, a rangeerror is thrown. exceptions rangeerror the property has been given a value that is outside the accepted range. ...
...if the value is set to less than 0, a rangeerror is thrown. exceptions rangeerror the property has been given a value that is outside the accepted range. ...
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RTCPeerConnection.addTrack() - Web APIs
ks if no streams are specified, then the track is streamless. this is perfectly acceptable, although it will be up to the remote peer to decide what stream to insert the track into, if any. this is a very common way to use addtrack() when building many types of simple applications, where only one stream is needed. for example, if all you're sharing with the remote peer is a single stream with an audio track and a video track, you don't need to deal with managing what track is in what stream, so you might as well just let the transceiver handle it for you. here's an example showing a function that uses getusermedia() to obtain a stream from a user's camera and microphone, then adds each track from the stream to the peer connection, without specifying a stream for each track: async opencall(pc...
...s well just let the transceiver handle it for you. here's an example showing a function that uses getusermedia() to obtain a stream from a user's camera and microphone, then adds each track from the stream to the peer connection, without specifying a stream for each track: async opencall(pc) { const gumstream = await navigator.mediadevices.getusermedia( {video: true, audio: true}); for (const track of gumstream.gettracks()) { pc.addtrack(track); } } the result is a set of tracks being sent to the remote peer, with no stream associations. the handler for the track event on the remote peer will be responsible for determining what stream to add each track to, even if that means simply adding them all to the same stream. the ontrack handler might look like thi...
... includes things like changes to the transceiver's direction and tracks being halted using removetrack(). for example, consider this function that an application might use to begin streaming a device's camera and microphone input over an rtcpeerconnection to a remote peer: async opencall(pc) { const gumstream = await navigator.mediadevices.getusermedia( {video: true, audio: true}); for (const track of gumstream.gettracks()) { pc.addtrack(track, gumstream); } } the remote peer might then use a track event handler that looks like this: pc.ontrack = ({streams: [stream]} => videoelem.srcobject = stream; this sets the video element's current stream to the one that contains the track that's been added to the connection. reused senders this method may return ...
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WaveShaperNode - Web APIs
the waveshapernode interface represents a non-linear distorter. it is an audionode that uses a curve to apply a wave shaping distortion to the signal. beside obvious distortion effects, it is often used to add a warm feeling to the signal. a waveshapernode always has exactly one input and one output. number of inputs 1 number of outputs 1 channel count mode "max" channel count 2 (not used in the default count mode) channel interpretation "speakers" constructor waveshapernode() creates a new instance of an waveshapernode object. properties inherits properties from its parent, audionode. waveshapernode.curve is a float32array of numbers describing the distortion to apply. waveshapernode.oversample is an enumerate...
...for applied examples/information, check out our voice-change-o-matic demo (see app.js for relevant code). ...
...for applied examples/information, check out our voice-change-o-matic demo (see app.js for relevant code). note: sigmoid functions are commonly used for distortion curves because of their natural properties. their s-shape, for instance, helps create a smoother sounding result. ...
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Signaling and video calling - Web APIs
webrtc allows real-time, peer-to-peer, media exchange between two devices. a connection is established through a discovery and negotiation process called signaling. this tutorial will guide you through building a two-way video-call. webrtc is a fully peer-to-peer technology for the real-time exchange of audio, video, and data, with one central caveat. a form of discovery and media format negotiation must take place, as discussed elsewhere, in order for two devices on different networks to locate one another. this process is called signaling and involves both devices connecting to a third, mutually agreed-upon server. through this third server, the two devices can locate one another, and exchange negotiation messages. in this article, we will further enhance the websocket chat first created a...
...<video> element with the id "received_video" will present video received from the connected user. we specify the autoplay attribute, ensuring once the video starts arriving, it immediately plays. this removes any need to explicitly handle playback in our code. the "local_video" <video> element presents a preview of the user's camera; specifiying the muted attribute, as we don't need to hear local audio in this preview panel. finally, the "hangup-button" <button>, to disconnect from a call, is defined and configured to start disabled (setting this as our default for when no call is connected) and apply the function hangupcall() on click. this function's role is to close the call, and send a signalling server notification to the other peer, requesting it also close. the javascript code we'll d...
...andler for the click event on the list item, that clicking on a user name calls our invite() method, which we'll look at in the next section. finally, we append the new item to the <ul> that contains all of the user names. starting a call when the user clicks on a username they want to call, the invite() function is invoked as the event handler for that click event: var mediaconstraints = { audio: true, // we want an audio track video: true // ...and we want a video track }; function invite(evt) { if (mypeerconnection) { alert("you can't start a call because you already have one open!"); } else { var clickedusername = evt.target.textcontent; if (clickedusername === myusername) { alert("i'm afraid i can't let you talk to yourself. that would be weird."); ret...
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Configuring servers for Ogg media - HTTP
html <audio> and <video> elements allow media presentation without the need for the user to install any plug-ins or other software to do so. this guide covers a few server configuration changes that may be necessary for your web server to correctly serve ogg media files. this information may also be useful if you encounter other media types your server isn't already configured to recognize. serve media with the correct mime type *.ogg and *.ogv files containing video (possibly with an audio track as well, of course), should be served with the video/ogg mime type. *.oga and *.ogg files containing only audio should be served with the audio/ogg mime type. ...
...ou encounter other media types your server isn't already configured to recognize. serve media with the correct mime type *.ogg and *.ogv files containing video (possibly with an audio track as well, of course), should be served with the video/ogg mime type. *.oga and *.ogg files containing only audio should be served with the audio/ogg mime type. if you don't know whether the ogg file contains audio or video, you can serve it with the mime type application/ogg, and the browser will treat it as a video file. ...
...dio/ogg mime type. if you don't know whether the ogg file contains audio or video, you can serve it with the mime type application/ogg, and the browser will treat it as a video file. most servers don't by default serve ogg media with the correct mime types, so you'll likely need to add the appropriate configuration for this. for apache, you can add the following to your configuration: addtype audio/ogg .oga addtype video/ogg .ogv addtype application/ogg .ogg you can find specific information about possible media file types and the codecs used within them in our comprehensive guide to media types and formats on the web. in particular, the article on media container formats will be especially helpful when configuring serers to host media properly. handle http 1.1 byte range requests correc...
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Index - Game development
m flexibility to optimize the compilation process. in gecko, async compilation allows the javascript engine to compile the asm.js off the main thread when the game is loading and cache the generated machine code so that the game doesn't need to be compiled on subsequent loads (starting in firefox 28). to see the difference, toggle javascript.options.parallel_parsing in about:config. ...
...e the compilation process. in gecko, async compilation allows the javascript engine to compile the asm.js off the main thread when the game is loading and cache the generated machine code so that the game doesn't need to be compiled on subsequent loads (starting in firefox 28). to see the difference, toggle javascript.options.parallel_parsing in about:config. ...
...process. in gecko, async compilation allows the javascript engine to compile the asm.js off the main thread when the game is loading and cache the generated machine code so that the game doesn't need to be compiled on subsequent loads (starting in firefox 28). to see the difference, toggle javascript.options.parallel_parsing in about:config. 26 audio for web games audio, games, web audio api, audio sprites, spatialization, syncing tracks audio is an important part of any game; it adds feedback and atmosphere. web-based audio is maturing fast, but there are still many browser differences to navigate. we often need to decide which audio parts are essential to our games' experience and which are nice to have but not essential, and devise a strategy accordingly. this article pr...
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Introduction to game development for the Web - Game development
ascript just-in-time compiler technology and new apis, you can build games that run in the browser (or on html5-powered devices) without making compromises. the html5 game platform you can truly think of the web as a better target platform for your game. ...
...in-time compiler technology and new apis, you can build games that run in the browser (or on html5-powered devices) without making compromises. the html5 game platform you can truly think of the web as a better target platform for your game. ...
...ame take over the entire screen, thereby immersing the player in action. gamepad api if you want your users to be able to use gamepads or other game controllers to work your game, you'll need this api. html and css together, these two technologies let you build, style, and lay out your game's user interface. part of html is the <canvas> element, which provides one way to do 2d graphics. html audio the <audio> element lets you easily play simple sound effects and music. if your needs are more involved, check out the web audio api for real audio processing power! indexeddb a powerful data storage api for maintaining user data on their own computer or device. a great way to save game state and other information locally so it doesn't have to be downloaded every time it's needed. also useful...
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HTML Cheatsheet - Learn web development
sed to format the date legibly for the user, such as: published on 23-05-2020 text displayed in code format <p>this text is in normal format.</p> <code>this text is in code format.</code> <pre>this text is in predefined format.</pre> this text is in normal format. this text is in code format. this text is in predefined format. ...
...mat the date legibly for the user, such as: published on 23-05-2020 text displayed in code format <p>this text is in normal format.</p> <code>this text is in code format.</code> <pre>this text is in predefined format.</pre> this text is in normal format. this text is in code format. this text is in predefined format. ...
...is in normal format.</p> <code>this text is in code format.</code> <pre>this text is in predefined format.</pre> this text is in normal format. this text is in code format. this text is in predefined format. embedded audio <audio controls="controls" src="https://udn.realityripple.com/samples/b7/193cb038d0.mp3">your browser does not support the html5 audio element.</audio> your browser does not support the html5 audio element. ...
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Handling common JavaScript problems - Learn web development
ll be disallowed that would otherwise be acceptable. it is arguably a good idea to use strict mode, as it makes for better, more efficient code, however it has limited/patchy support across browsers (see strict mode in browsers). typed arrays allow javascript code to access and manipulate raw binary data, which is necessary as browser apis for example start to manipulate streams of raw video and audio data. these are available in ie10 and above, and all modern browsers. there are also many new apis appearing in recent browsers, which don't work in older browsers, for example: indexeddb api, web storage api, and others for storing website data on the client-side. web workers api for running javascript in a separate thread, helping to improve performance. webgl api for real 3d graphics. ...
...a. these are available in ie10 and above, and all modern browsers. there are also many new apis appearing in recent browsers, which don't work in older browsers, for example: indexeddb api, web storage api, and others for storing website data on the client-side. web workers api for running javascript in a separate thread, helping to improve performance. webgl api for real 3d graphics. web audio api for advanced audio manipulation. webrtc api for multi-person, real-time video/audio connectivity (e.g. video conferencing). there are a few strategies for handling incompatibilities between browsers relating to feature support; let's explore the most common ones. note: these strategies do not exist in separate silos — you can, of course combine them as needed. for example, you could us...
...efixes too, although javascript uses camel case rather than hyphenation like css. for example, if a prefix was being used on a new shint api object called object: mozilla would use mozobject chrome/opera/safari would use webkitobject microsoft would use msobject here's an example, taken from our violent-theremin demo (see source code), which uses a combination of the canvas api and the web audio api to create a fun (and noisy) drawing tool: const audiocontext = window.audiocontext || window.webkitaudiocontext; const audioctx = new audiocontext(); in the case of the web audio api, the key entry points to using the api were supported in chrome/opera via webkit prefixed versions (they now support the unprefixed versions). the easy way to get around this situation is to create a new versio...
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AnalyserNode.getByteFrequencyData() - Web APIs
cibel value for a specific frequency. the frequencies are spread linearly from 0 to 1/2 of the sample rate. for example, for 48000 sample rate, the last item of the array will represent the decibel value for 24000 hz. if the array has fewer elements than the analysernode.frequencybincount, excess elements are dropped. if it has more elements than needed, excess elements are ignored. ...
... a specific frequency. the frequencies are spread linearly from 0 to 1/2 of the sample rate. for example, for 48000 sample rate, the last item of the array will represent the decibel value for 24000 hz. if the array has fewer elements than the analysernode.frequencybincount, excess elements are dropped. if it has more elements than needed, excess elements are ignored. syntax var audioctx = new audiocontext(); var analyser = audioctx.createanalyser(); var dataarray = new uint8array(analyser.frequencybincount); // uint8array should be the same length as the frequencybincount void analyser.getbytefrequencydata(dataarray); // fill the uint8array with data returned from getbytefrequencydata() parameters array the uint8array that the frequency domain data will be copied to. for any sample...
...ta() parameters array the uint8array that the frequency domain data will be copied to. for any sample which is silent, the value is -infinity. if the array has fewer elements than the analysernode.frequencybincount, excess elements are dropped. if it has more elements than needed, excess elements are ignored. return value none. ...
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AnalyserNode.getByteTimeDomainData() - Web APIs
the getbytetimedomaindata() method of the analysernode interface copies the current waveform, or time-domain, data into a uint8array (unsigned byte array) passed into it. if the array has fewer elements than the analysernode.fftsize, excess elements are dropped. if it has more elements than needed, excess elements are ignored. syntax const audioctx = new audiocontext(); const analyser = audioctx.createanalyser(); const dataarray = new uint8array(analyser.fftsize); // uint8array should be the same length as the fftsize analyser.getbytetimedomaindata(dataarray); // fill the uint8array with data returned from getbytetimedomaindata() parameters array the uint8array that the time domain data will be copied to. if the array has fewer elements than the analysernode.fftsize, excess elements ...
...); // fill the uint8array with data returned from getbytetimedomaindata() parameters array the uint8array that the time domain data will be copied to. if the array has fewer elements than the analysernode.fftsize, excess elements are dropped. if it has more elements than needed, excess elements are ignored. ...
... fewer elements than the analysernode.fftsize, excess elements are dropped. if it has more elements than needed, excess elements are ignored. return value void | none example the following example shows basic usage of an audiocontext to create an analysernode, then requestanimationframe and <canvas> to collect time domain data repeatedly and draw an "oscilloscope style" output of the current audio input. for more complete applied examples/information, check out our voice-change-o-matic demo (see app.js lines 128–205 for relevant code). const audioctx = new (window.audiocontext || window.webkitaudiocontext)(); const analyser = audioctx.createanalyser(); ... analyser.fftsize = 2048; const bufferlength = analyser.fftsize; const dataarray = new uint8array(bufferlength); analyser.getbyte...
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Body.arrayBuffer() - Web APIs
it to completion. it returns a promise that resolves with an arraybuffer. syntax response.arraybuffer().then(function(buffer) { // do something with buffer }); parameters none. return value a promise that resolves with an arraybuffer. examples playing music in our fetch array buffer live, we have a play button. when pressed, the getdata() function is run. note that before playing full audio file will be downloaded. if you need to play ogg during downloading (stream it) - consider htmlaudioelement: new audio("music.ogg").play(); in getdata() we create a new request using the request() constructor, then use it to fetch an ogg music track. we also use audiocontext.createbuffersource to create an audio buffer source. when the fetch is successful, we read an arraybuffer out of the res...
...to play ogg during downloading (stream it) - consider htmlaudioelement: new audio("music.ogg").play(); in getdata() we create a new request using the request() constructor, then use it to fetch an ogg music track. we also use audiocontext.createbuffersource to create an audio buffer source. when the fetch is successful, we read an arraybuffer out of the response using arraybuffer(), decode the audio data using audiocontext.decodeaudiodata, set the decoded data as the audio buffer source's buffer (source.buffer), then connect the source up to the audiocontext.destination. ...
...iocontext.createbuffersource to create an audio buffer source. when the fetch is successful, we read an arraybuffer out of the response using arraybuffer(), decode the audio data using audiocontext.decodeaudiodata, set the decoded data as the audio buffer source's buffer (source.buffer), then connect the source up to the audiocontext.destination. ...
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ConvolverNode() - Web APIs
disablenormalization: a boolean controlling whether the impulse response from the buffer will be scaled by an equal-power normalization, or not. the default is 'false'. return value a new convolvernode object instance. ...
... disablenormalization: a boolean controlling whether the impulse response from the buffer will be scaled by an equal-power normalization, or not. the default is 'false'. return value a new convolvernode object instance. ...
... disablenormalization: a boolean controlling whether the impulse response from the buffer will be scaled by an equal-power normalization, or not. the default is 'false'. return value a new convolvernode object instance. ...
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ConvolverNode.buffer - Web APIs
the buffer property of the convolvernode interface represents a mono, stereo, or 4-channel audiobuffer containing the (possibly multichannel) impulse response used by the convolvernode to create the reverb effect. this is normally a simple recording of as-close-to-an-impulse as can be found in the space you want to model. for example, if you want to model the reverb in your bathroom, you might set up a microphone near the door to record the sound of a balloon pop or synthesized impulse from the sink. that audio recording could then be used as the buffer. this audiobuffer must have the same sample-rate as the audiocontext or an exception will be thrown. at the time when this attribute is set, the buffer and the state of the attribute will be used to configure the convolvernode with this impu...
...that audio recording could then be used as the buffer. this audiobuffer must have the same sample-rate as the audiocontext or an exception will be thrown. at the time when this attribute is set, the buffer and the state of the attribute will be used to configure the convolvernode with this impulse response having the given normalization. the initial value of this attribute is null. syntax var audioctx = new audiocontext(); var convolver = audioctx.createconvolver(); convolver.buffer = myaudiobuffer; value an audiobuffer. example var audioctx = new (window.audiocontext || window.webkitaudiocontext)(); var convolver = audioctx.createconvolver(); ... ...
...node with this impulse response having the given normalization. the initial value of this attribute is null. syntax var audioctx = new audiocontext(); var convolver = audioctx.createconvolver(); convolver.buffer = myaudiobuffer; value an audiobuffer. example var audioctx = new (window.audiocontext || window.webkitaudiocontext)(); var convolver = audioctx.createconvolver(); ... ...
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ConvolverNode - Web APIs
the convolvernode interface is an audionode that performs a linear convolution on a given audiobuffer, often used to achieve a reverb effect. a convolvernode always has exactly one input and one output. note: for more information on the theory behind linear convolution, see the convolution article on wikipedia. number of inputs 1 number of outputs 1 channel count mode "clamped-max" channel count 1, 2, or 4 channel interpretation "speakers" constructor convolvernode() creates a new convolvernode object instance. properties inherits properties from its parent, audionode. ...
...pedia. number of inputs 1 number of outputs 1 channel count mode "clamped-max" channel count 1, 2, or 4 channel interpretation "speakers" constructor convolvernode() creates a new convolvernode object instance. properties inherits properties from its parent, audionode. convolvernode.buffer a mono, stereo, or 4-channel audiobuffer containing the (possibly multichannel) impulse response used by the convolvernode to create the reverb effect. convolvernode.normalize a boolean that controls whether the impulse response from the buffer will be scaled by an equal-power normalization when the buffer attribute is set, or not. methods no specific method; inherits methods from its parent, audionode. convolvernode example...
...pulse response used by the convolvernode to create the reverb effect. convolvernode.normalize a boolean that controls whether the impulse response from the buffer will be scaled by an equal-power normalization when the buffer attribute is set, or not. methods no specific method; inherits methods from its parent, audionode. convolvernode example the following example shows basic usage of an audiocontext to create a convolver node. ...
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KeyboardEvent: code values - Web APIs
ut" 0xe018 "unidentified" "copy" 0xe019 "mediatracknext" "mediatracknext" 0xe01a, 0xe01b "unidentified" "" 0xe01c "numpadenter" "numpadenter" 0xe01d "controlright" "controlright" 0xe01e "unidentified" "launchmail" 0xe01f "unidentified" "" 0xe020 "audiovolumemute" "audiovolumemute" 0xe021 "launchapp2" "" 0xe022 "mediaplaypause" "mediaplaypause" 0xe023 "unidentified" "" 0xe024 "mediastop" "mediastop" 0xe025 ～ 0xe02b "unidentified" "" 0xe02c "unidentified" "eject" 0xe02d "unidentified" "" 0xe02e...
..." "audiovolumemute" 0xe021 "launchapp2" "" 0xe022 "mediaplaypause" "mediaplaypause" 0xe023 "unidentified" "" 0xe024 "mediastop" "mediastop" 0xe025 ～ 0xe02b "unidentified" "" 0xe02c "unidentified" "eject" 0xe02d "unidentified" "" 0xe02e "audiovolumedown" "volumedown" (was "volumedown" until chrome 50) 0xe02f "unidentified" "" 0xe030 "audiovolumeup" "volumeup" (was "volumeup" until chrome 50) 0xe031 "unidentified" "" 0xe032 "browserhome" "browserhome" 0xe033, 0xe034 "unidentified" "" 0xe035 "numpaddivide" ...
...ally) "" (no events fired actually) kvk_f17 (0x40) "f17" "f17" kvk_ansi_keypaddecimal (0x41) "numpaddecimal" "numpaddecimal" kvk_ansi_keypadmultiply (0x43) "numpadmultiply" "numpadmultiply" kvk_ansi_keypadplus (0x45) "numpadadd" "numpadadd" kvk_ansi_keypadclear (0x47) "numlock" "numlock" kvk_volumeup (0x48) "audiovolumeup" (was "volumeup" until firefox 48) "audiovolumeup" (was "volumeup" until chrome 50) kvk_volumedown (0x49) "audiovolumedown" (was "volumedown" until firefox 49) "audiovolumedown" (was "volumedown" until chrome 50) kvk_mute (0x4a) "audiovolumemute" (was "volumemute" until firefox 49) "audiovolumemute" (was "volumemute" until chrome 50) kvk_ansi_keyp...
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MediaRecorder - Web APIs
the mediarecorder interface of the mediastream recording api provides functionality to easily record media. it is created using the mediarecorder() constructor. constructor mediarecorder() creates a new mediarecorder object, given a mediastream to record. options are available to do things like set the container's mime type (such as "video/webm" or "video/mp4") and the bit rates of the audio and video tracks or a single overall bit rate. properties mediarecorder.mimetype read only returns the mime type that was selected as the recording container for the mediarecorder object when it was created. mediarecorder.state read only returns the current state of the mediarecorder object (inactive, recording, or paused.) mediarecorder.stream read only returns the stream that was passed...
...t state of the mediarecorder object (inactive, recording, or paused.) mediarecorder.stream read only returns the stream that was passed into the constructor when the mediarecorder was created. mediarecorder.ignoremutedmedia indicates whether the mediarecorder instance will record input when the input mediastreamtrack is muted. if this attribute is false, mediarecorder will record silence for audio and black frames for video. the default is false. mediarecorder.videobitspersecond read only returns the video encoding bit rate in use. this may differ from the bit rate specified in the constructor (if it was provided). mediarecorder.audiobitspersecond read only returns the audio encoding bit rate in use. this may differ from the bit rate specified in the constructor (if it was provided).
...n to these events using addeventlistener() or by assigning an event listener to the oneventname property of this interface. error fired when an error occurs: for example because recording wasn't allowed or was attempted using an unsupported codec. also available via the onerror property. ...
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OscillatorNode.start() - Web APIs
the start method of the oscillatornode interface specifies the exact time to start playing the tone. its parameter is optional and default to 0. ...
...ying oscillator at the point in time specified by when parameters when optional an optional double representing the time (in seconds) when the oscillator should start, in the same coordinate system as audiocontext's currenttime attribute. if a value is not included or is less than currenttime, the oscillator starts playing immediately. example the following example shows basic usage of an audiocontext to create an oscillator node. for an applied example, check out our violent theremin demo (see app.js for relevant code). ...
...cillator should start, in the same coordinate system as audiocontext's currenttime attribute. if a value is not included or is less than currenttime, the oscillator starts playing immediately. example the following example shows basic usage of an audiocontext to create an oscillator node. for an applied example, check out our violent theremin demo (see app.js for relevant code). ...
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OscillatorNode.stop() - Web APIs
the stop method of the oscillatornode interface specifies the time to stop playing the tone. its parameter is optional and defaults to 0. syntax oscillator.stop(when); // stop playing oscillator at when parameters when optional an optional double representing the audio context time when the oscillator should stop. if a value is not included, it defaults to 0. if the time is equal to or before the current audio context time, the oscillator will stop playing immediately. ...
... syntax oscillator.stop(when); // stop playing oscillator at when parameters when optional an optional double representing the audio context time when the oscillator should stop. if a value is not included, it defaults to 0. if the time is equal to or before the current audio context time, the oscillator will stop playing immediately. example the following example shows basic usage of an audiocontext to create an oscillator node. for an applied example, check out our violent theremin demo (see app.js for relevant code). ...
...e when the oscillator should stop. if a value is not included, it defaults to 0. if the time is equal to or before the current audio context time, the oscillator will stop playing immediately. example the following example shows basic usage of an audiocontext to create an oscillator node. for an applied example, check out our violent theremin demo (see app.js for relevant code). ...
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PannerNode.orientationY - Web APIs
the orientationy property of the pannernode interface indicates the y (vertical) component of the direction the audio source is facing, in 3d cartesian coordinate space. the complete vector is defined by the position of the audio source, given as (positionx, positiony, positionz), and the orientation of the audio source (that is, the direction in which it's facing), given as (orientationx, orientationy, orientationz). depending on the directionality of the sound (as specified using the attributes coneinnerangle, coneouterangle, and codeoutergain), the orientation of the sound may alter the perceived volume of the sound as it's being played. ...
...en as (orientationx, orientationy, orientationz). depending on the directionality of the sound (as specified using the attributes coneinnerangle, coneouterangle, and codeoutergain), the orientation of the sound may alter the perceived volume of the sound as it's being played. if the sound is pointing toward the listener, it will be louder than if the sound is pointed away from the listener. the audioparam contained by this property is read only; however, you can still change the value of the parameter by assigning a new value to its audioparam.value property. syntax var orientationy = pannernode.orientationy; pannernode.orientationy.value = neworientationy; value an audioparam whose value is the y component of the direction the audio source is facing, in 3d cartesian coordinate space. ...
... offset the angle so 0 points towards the listener const radians = (degrees - 90) * (math.pi / 180); // using cosine and sine here ensures the output values are always normalised // i.e. ...
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PannerNode.orientationZ - Web APIs
the orientationz property of the pannernode interface indicates the z (depth) component of the direction the audio source is facing, in 3d cartesian coordinate space. the complete vector is defined by the position of the audio source, given as (positionx, positiony, positionz), and the orientation of the audio source (that is, the direction in which it's facing), given as (orientationx, orientationy, orientationz). depending on the directionality of the sound (as specified using the attributes coneinnerangle, coneouterangle, and codeoutergain), the orientation of the sound may alter the perceived volume of the sound as it's being played. ...
...en as (orientationx, orientationy, orientationz). depending on the directionality of the sound (as specified using the attributes coneinnerangle, coneouterangle, and codeoutergain), the orientation of the sound may alter the perceived volume of the sound as it's being played. if the sound is pointing toward the listener, it will be louder than if the sound is pointed away from the listener. the audioparam contained by this property is read only; however, you can still change the value of the parameter by assigning a new value to its audioparam.value property. syntax var orientationz = pannernode.orientationz; pannernode.orientationz.value = neworientationz; value an audioparam whose value is the z component of the direction the audio source is facing, in 3d cartesian coordinate space. ...
... offset the angle so 0 points towards the listener const radians = (degrees - 90) * (math.pi / 180); // using cosine and sine here ensures the output values are always normalised // i.e. ...
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RTCStatsReport - Web APIs
information about the state of the remote peers (such as round-trip time) is unavailable, since those values can't be computed without knowing about the other peers' states. peer-connection a rtcpeerconnectionstats object provides statistics related to the overall peer connection's rtcpeerconnection. receiver provides statistics about a specific rtcrtpreceiver. the statistics object is an rtcaudioreceiverstats object if kind is audio; if kind is video, the object is an rtcvideoreceiverstats object. remote-candidate the report is an rtcicecandidatestats object containing statistics about the remote candidate's rtcicetransport. this may include information such as the type of network, the protocol, the url, the type of relay being used, and so forth. remote-inbound-rtp the report is an r...
...se this information to do things like determine how well the remote peer is receiving data. remote-outbound-rtp the report is an rtcremoteoutboundrtpstreamstats object that contains statistics about your inbound rtp (inbound-rtp) stream, but from the perspective of the remote peer. sender an object containing statistics about the rtcrtpsender for a stream on the rtcpeerconnection. ...
...ike determine how well the remote peer is receiving data. remote-outbound-rtp the report is an rtcremoteoutboundrtpstreamstats object that contains statistics about your inbound rtp (inbound-rtp) stream, but from the perspective of the remote peer. sender an object containing statistics about the rtcrtpsender for a stream on the rtcpeerconnection. ...
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RTCStatsType - Web APIs
information about the state of the remote peers (such as round-trip time) is unavailable, since those values can't be computed without knowing about the other peers' states. peer-connection a rtcpeerconnectionstats object provides statistics related to the overall peer connection's rtcpeerconnection. receiver provides statistics about a specific rtcrtpreceiver. the statistics object is an rtcaudioreceiverstats object if kind is audio; if kind is video, the object is an rtcvideoreceiverstats object. remote-candidate the report is an rtcicecandidatestats object containing statistics about the remote candidate's rtcicetransport. this may include information such as the type of network, the protocol, the url, the type of relay being used, and so forth. remote-inbound-rtp the report is an r...
...se this information to do things like determine how well the remote peer is receiving data. remote-outbound-rtp the report is an rtcremoteoutboundrtpstreamstats object that contains statistics about your inbound rtp (inbound-rtp) stream, but from the perspective of the remote peer. sender an object containing statistics about the rtcrtpsender for a stream on the rtcpeerconnection. ...
...ike determine how well the remote peer is receiving data. remote-outbound-rtp the report is an rtcremoteoutboundrtpstreamstats object that contains statistics about your inbound rtp (inbound-rtp) stream, but from the perspective of the remote peer. sender an object containing statistics about the rtcrtpsender for a stream on the rtcpeerconnection. ...
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SpeechRecognition - Web APIs
the speechrecognition interface of the web speech api is the controller interface for the recognition service; this also handles the speechrecognitionevent sent from the recognition service. note: on some browsers, like chrome, using speech recognition on a web page involves a server-based recognition engine. your audio is sent to a web service for recognition processing, so it won't work offline. constructor speechrecognition.speechrecognition() creates a new speechrecognition object. properties speechrecognition also inherits properties from its parent interface, eventtarget. speechrecognition.grammars returns and sets a collection of speechgrammar objects that represent the grammars that will be understood by the current speechrecognition. speechrecognition.lang returns a...
...echrecognition.serviceuri specifies the location of the speech recognition service used by the current speechrecognition to handle the actual recognition. the default is the user agent's default speech service. methods speechrecognition also inherits methods from its parent interface, eventtarget. ...
...on. the default is the user agent's default speech service. methods speechrecognition also inherits methods from its parent interface, eventtarget. speechrecognition.abort() stops the speech recognition service from listening to incoming audio, and doesn't attempt to return a speechrecognitionresult. speechrecognition.start() starts the speech recognition service listening to incoming audio with intent to recognize grammars associated with the current speechrecognition. speechrecognition.stop() stops the speech recognition service from listening to incoming audio, and attempts to return a speechrecognitionresult using the audio captured so far. events listen to these events using addeventlistener() or by assigning an event listener to the oneventname property of this interface.
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TrackEvent - Web APIs
ement; these events are addtrack and removetrack. it's important not to confuse trackevent with the rtctrackevent interface, which is used for tracks which are part of an rtcpeerconnection. events based on trackevent are always sent to one of the media track list types: events involving video tracks are always sent to the videotracklist found in htmlmediaelement.videotracks events involving audio tracks are always sent to the audiotracklist specified in htmlmediaelement.audiotracks events affecting text tracks are sent to the texttracklist object indicated by htmlmediaelement.texttracks. constructor trackevent() creates and initializes a new trackevent object with the event type specified, as well as optional additional properties. properties trackevent is based on event, so ...
...structor trackevent() creates and initializes a new trackevent object with the event type specified, as well as optional additional properties. properties trackevent is based on event, so properties of event are also available on trackevent objects. track read only the dom track object the event is in reference to. if not null, this is always an object of one of the media track types: audiotrack, videotrack, or texttrack). methods trackevent has no methods of its own; however, it is based on event, so it provides the methods available on event objects. example this example sets up a function, handletrackevent(), which is callled for any addtrack or removetrack event on the first <video> element found in the document. var videoelem = document.queryselector("video"); videoelem.
...nt objects. example this example sets up a function, handletrackevent(), which is callled for any addtrack or removetrack event on the first <video> element found in the document. ...
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WebRTC API - Web APIs
the set of standards that comprise webrtc makes it possible to share data and perform teleconferencing peer-to-peer, without requiring that the user installs plug-ins or any other third-party software. webrtc consists of several interrelated apis and protocols which work together to achieve this. ...
...e webrtc development process easier, with more broadly compatible results. the library is also available as an npm package. to learn more about adapter.js, see improving compatibility using webrtc adapter.js. webrtc concepts and usage webrtc serves multiple purposes; together with the media capture and streams api, they provide powerful multimedia capabilities to the web, including support for audio and video conferencing, file exchange, screen sharing, identity management, and interfacing with legacy telephone systems including support for sending dtmf (touch-tone dialing) signals. connections between peers can be made without requiring any special drivers or plug-ins, and can often be made without any intermediary servers. connections between two peers are represented by the rtcpeerconnec...
...erface. once a connection has been established and opened using rtcpeerconnection, media streams (mediastreams) and/or data channels (rtcdatachannels) can be added to the connection. media streams can consist of any number of tracks of media information; tracks, which are represented by objects based on the mediastreamtrack interface, may contain one of a number of types of media data, including audio, video, and text (such as subtitles or even chapter names). most streams consist of at least one audio track and likely also a video track, and can be used to send and receive both live media or stored media information (such as a streamed movie). you can also use the connection between two peers to exchange arbitrary binary data using the rtcdatachannel interface. this can be used for back-chan...
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<source>: The Media or Image Source element - HTML: Hypertext Markup Language
it is an empty element, meaning that it has no content and does not have a closing tag. it is commonly used to offer the same media content in multiple file formats in order to provide compatibility with a broad range of browsers given their differing support for image file formats and media file formats. the source for this interactive example is stored in a github repository. if you'd like to contribute to the interactive examples project, please clone https://github.com/mdn/interactive-examples and send us a pull request. content categories none. ...
...thub repository. if you'd like to contribute to the interactive examples project, please clone https://github.com/mdn/interactive-examples and send us a pull request. content categories none. permitted content none, it is an empty element. tag omission it must have a start tag, but must not have an end tag. permitted parents a media element—<audio> or <video>—and it must be placed before any flow content or <track> element. a <picture> element, and it must be placed before the <img> element. implicit aria role no corresponding role permitted aria roles no role permitted dom interface htmlsourceelement attributes this element includes the global attributes. media media query of the r...
...ed by the browser to determine, before laying the page out, which image defined in srcset to use. please note that sizes will have its effect only if width dimension descriptors are provided with srcset instead of pixel ratio values (200w instead of 2x for example). the sizes attribute has an effect only when the <source> element is the direct child of a <picture> element. src required for <audio> and <video>, address of the media resource. the value of this attribute is ignored when the <source> element is placed inside a <picture> element. srcset a list of one or more strings separated by commas indicating a set of possible images represented by the source for the browser to use. each string is composed of: one url specifying an image. a width descriptor, which consists of a ...
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HTML elements reference - HTML: Hypertext Markup Language
e name of a variable in a mathematical expression or a programming context. <wbr> the html <wbr> element represents a word break opportunity—a position within text where the browser may optionally break a line, though its line-breaking rules would not otherwise create a break at that location. ...
...les would not otherwise create a break at that location. image and multimedia html supports various multimedia resources such as images, audio, and video. element description <area> the html <area> element defines a hot-spot region on an image, and optionally associates it with a hypertext link. this element is used only within a <map> element. <audio> the html <audio> element is used to embed sound content in documents. it may contain one or more audio sources, represented using the src attribute or the <source> element: the browser will choose the most suitable one. it can also be the destination for streamed media, using a mediastream. <img> the html <img> element embeds an image into the document. <map> the h...
...wser will choose the most suitable one. it can also be the destination for streamed media, using a mediastream. <img> the html <img> element embeds an image into the document. <map> the html <map> element is used with <area> elements to define an image map (a clickable link area). <track> the html <track> element is used as a child of the media elements, <audio> and <video>. it lets you specify timed text tracks (or time-based data), for example to automatically handle subtitles. <video> the html video element (<video>) embeds a media player which supports video playback into the document. you can use <video> for audio content as well, but the <audio> element may provide a more appropriate user experience. embedded content in addi...
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HTTP Index - HTTP
me types (iana media types) content-type, guide, http, mime types, meta, request header, response header, application/javascript, application/json, application/xml a media type (also known as a multipurpose internet mail extensions or mime type) is a standard that indicates the nature and format of a document, file, or assortment of bytes. 10 incomplete list of mime types audio, file types, files, http, mime, mime types, php, reference, text, types, video here is a list of mime types, associated by type of documents, ordered by their common extensions. 11 resource urls guide, http, intermediate, resource resource urls, urls prefixed with the resource: scheme, are used by firefox and firefox browser extensions to load resources internally, but ...
...cific browsers. 13 compression in http guide, http, compression compression is an important way to increase the performance of a web site. for some documents, size reduction of up to 70% lowers the bandwidth capacity needs. over the years, algorithms also got more efficient, and new ones are supported by clients and servers. 14 configuring servers for ogg media audio, http, media, ogg, video this guide covers a few server configuration changes that may be necessary for your web server to correctly serve ogg media files. 15 connection management in http/1.x connection management, guide, http, networking, performance, webmechanics connection management is a key topic in http: opening and maintaining connections largely impacts the per...
...nce, security, manifest-src, source the http content-security-policy: manifest-src directive specifies which manifest can be applied to the resource. 88 csp: media-src csp, content-security-policy, directive, http, media, reference, security, media-src, source the http content-security-policy (csp) media-src directive specifies valid sources for loading media using the <audio> and <video> elements. 89 csp: navigate-to csp, content-security-policy, directive, http, navigation, reference, security the http content-security-policy (csp) navigate-to directive restricts the urls to which a document can initiate navigations by any means including <form> (if form-action is not specified), <a>, window.location, window.open, etc. this is an enforcement on ...
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XUL accessibility guidelines - Archive of obsolete content
ibe keyboard shortcuts and any other accessibility considerations. users should be able to reference a complete description of all major functionality of an application. also provide details on how to use long or complex help documentation. alerts provide accessible alerts for presenting important information to the user. use scripting or the notificationbox element to flag alerts. avoid using audio or visual alerts alone to signal urgent events. users who have audio turned down or off and users who are deaf or hard of hearing may not be able to recognize audio only alerts. users with visual disabilities will not be able to observe alerts that are purely visual (unless such alerts are presented in functional text that is accessible to a screen reader). interactive elements avoid small targ...
...ively process what is happening in the application. testing human computer interaction ensure that help documentation is up to date. verify that alerts are displayed through appropriate xul elements. ensure the user is informed of any user based errors and that instruction and opportunity for reperforming error causing actions is allowed. ensure that users have control of response time. media audio informative audio files such as podcasts can be made accessible by supplying a word-for-word transcript. transcripts should identify the speakers and describe other relevant sounds such as laughing or singing. transcripts can be time consuming to generate, but is the only way to make audio content accessible. video a video file can be made accessible by adding synchronized captions. most media...
...aptions. most media delivery formats provide a means for displaying captions. video should also have descriptive transcripts. usually captioning and transcribing go hand in hand, once you have one its easy to produce the other. learn more captioning webaim article: web captioning overview webaim resource: captioning resources animation animation, movement, and audio can all be distracting to some users with attention disorders. provide a mechanism for turning media and movement on and off. flickering or flashing is not only just annoying to everyone, but at a rate more than 3 cycles per second it can cause a seizure in users with photosensitive epilepsy. if flickering or flashing is necessary then warn the user before it is displayed. testing media ensure...
...rror. the user has the opportunity and instruction to fix the error. no error alerts exist, or inadequate instruction is provided. response time user is informed of time limits and has control of response time when appropriate. user is not notified of time limits and/or has no control over response time when appropriate. media checkpoint pass fail audio transcripts are provided for audio tracks. no transcript provided. video video is captioned and a transcript is provided. no captions and/or transcripts provided. animation user has control over animation and is warned about flashing content. no control over animation or warnings about flashing exist. other checkpoint pass fail c...

Plug-in Development Overview - Gecko Plugin API Reference
the resource code for this language and character set combination is 040904e4. unix on unix, plug-ins are found according to the section how gecko finds plug-ins. to determine the mime types and fi...
...ould use the info.plist approach, and only use the following method as a supplement to support older browsers. if neither is present, gecko will check the plugins resources. 'str#' 128 should contain a list of mime types and file extensions in alternating strings. for example: str 128 mime type string 1 video/quicktime string 2 mov, moov string 3 audio/aiff string 4 aiff string 5 image/jpeg string 6 jpg several other optional strings may contain useful information about the plug-in. plug-ins must support 'str#' 128 but are not required to support any of these others: str#' 127 can contain a list of mime type descriptions corresponding to the types in 'str#' 128 . for example, this description list c...
... several other optional strings may contain useful information about the plug-in. plug-ins must support 'str#' 128 but are not required to support any of these others: str#' 127 can contain a list of mime type descriptions corresponding to the types in 'str#' 128 . for example, this description list corresponds to the types in the previous example: string 1: "quicktime video", string 4: "aiff audio", and string 5: "jpeg image format." str#' 126: string 1 can contain a descriptive message about the plug-in. this message, which is in html format, is displayed by the browser in its "about plug-ins" page. string 2 can contain the name of the plug-in, thus allowing the name the user sees to be different from the name of the file on disk. drawing a plug-in instance before drawing itself on...
...tus information. making plug-ins scriptable scriptable plug-ins are plug-ins that have been extended to provide methods that can be called from javascript and the dom when accessed through the object or embed element. consider the following example, where a media player plug-in can be controlled with an advancetonextsong() method called inside the script element: <object id="myplugin" type="audio/wav" data="music.wav"> </object> <script type="application/javascript"> var theplugin = document.getelementbyid('myplugin'); if (theplugin && theplugin.advancetonextsong) theplugin.advancetonextsong(); else alert("plugin not installed correctly"); </script> in the past, liveconnect and later xpconnect were used to make plug-ins scriptable. see the wikipedia article for hist...

Experimental features in Firefox
re the element's value changes. release channel version added enabled by default? nightly 74 no developer edition 74 no beta 74 no release 74 no preference name dom.input_events.beforeinput.enabled htmlmediaelement method: setsinkid() htmlmediaelement.setsinkid() allows you to set the sink id of an audio output device on an htmlmediaelement, thereby changing where the audio is being output. see bug 934425 for more details. release channel version added enabled by default? nightly 64 no developer edition 64 no beta 64 no release 64 no preference name media.setsinkid.enabled htmlmediaelement propertie...
...tput device on an htmlmediaelement, thereby changing where the audio is being output. see bug 934425 for more details. release channel version added enabled by default? nightly 64 no developer edition 64 no beta 64 no release 64 no preference name media.setsinkid.enabled htmlmediaelement properties: audiotracks and videotracks enabling this feature adds the htmlmediaelement.audiotracks and htmlmediaelement.videotracks properties to all html media elements. however, because firefox doesn't currently suport multiple audio and video tracks, the most common use cases for these properties don't work, so they're both disabled by default. see bug 1057233 for more details. release channel ve...
...autoplay property on a media element or by attempting to trigger playback from javascript code. the spec for this api is still being written. the value changes over time depending on what the user is doing, their preferences, and the state of the browser in general. potential values include allowed (autoplay is currently permitted), allowed-muted (autoplay is allowed but only with no—or muted—audio), and disallowed (autoplay is not allowed at this time). see bug 1506289 for more details. release channel version added enabled by default? nightly 66 no developer edition 66 no beta 66 no release 66 no preference name dom.media.autoplay.autoplay-policy-api geometryutils methods: convertpointfromno...
...1520690 for more details. release channel version added enabled by default? nightly 73 no developer edition 73 no beta 73 no release 73 no preference name layout.css.constructable-stylesheets.enabled webrtc and media the following experimental features include those found in the webrtc api, the web audio api, the media session api, the media source extensions api, the encrypted media extensions api, and the media capture and streams api. media session api the entire firefox implementation of the media session api is currently experimental. this api is used to customize the handling of media-related notifications, to manage events and data useful for presenting a user interface for managing medi...

Browser API
ction usage such as requestanimationframe. htmliframeelement.getvisible() indicates the current visibility state of the browser <iframe>. htmliframeelement.setactive() sets the current <iframe> as the active frame, which has an effect on how it is prioritized by the process manager. htmliframeelement.getactive() indicates whether the current browser <iframe> is the currently active frame. ...
... htmliframeelement.getmuted() indicates whether the browser <iframe> is currently muted. ...
... htmliframeelement.getmuted() indicates whether the browser <iframe> is currently muted. event-related methods in order to manage the browser <iframe>'s content, many new events were added (see below). ...
...g within the browser <iframe>. the following new events can be listened for: mozbrowseractivitydone sent when something inside the browser <iframe> triggers a web activity, and that web activity's message is consumed by the receiving app. mozbrowserasyncscroll sent when the browser <iframe> content scrolls (the apcz version.) see also mozbrowserscroll. note that this is obsolete. mozbrowseraudioplaybackchange sent when audio starts or stops playing within the browser <iframe> content. mozbrowsercaretstatechanged sent when the text selected inside the browser <iframe> content changes. mozbrowserclose sent when window.close() is called within a browser <iframe>. mozbrowsercontextmenu sent when a browser <iframe> tries to open a context menu. this allows handling of the <menuitem> e...

Plug-in Development Overview - Plugins
the resource code for this language and character set combination is 040904e4. unix on unix, plug-ins are found according to the section how gecko finds plug-ins. to determine the mime types and fi...
...ould use the info.plist approach, and only use the following method as a supplement to support older browsers. if neither is present, gecko will check the plugins resources. 'str#' 128 should contain a list of mime types and file extensions in alternating strings. for example: str 128 mime type string 1 video/quicktime string 2 mov, moov string 3 audio/aiff string 4 aiff string 5 image/jpeg string 6 jpg several other optional strings may contain useful information about the plug-in. plug-ins must support 'str#' 128 but are not required to support any of these others: str#' 127 can contain a list of mime type descriptions corresponding to the types in 'str#' 128 . for example, this description list c...
... several other optional strings may contain useful information about the plug-in. plug-ins must support 'str#' 128 but are not required to support any of these others: str#' 127 can contain a list of mime type descriptions corresponding to the types in 'str#' 128 . for example, this description list corresponds to the types in the previous example: string 1: "quicktime video", string 4: "aiff audio", and string 5: "jpeg image format." str#' 126: string 1 can contain a descriptive message about the plug-in. this message, which is in html format, is displayed by the browser in its "about plug-ins" page. string 2 can contain the name of the plug-in, thus allowing the name the user sees to be different from the name of the file on disk. drawing a plug-in instance before drawing itself on...
...tus information. making plug-ins scriptable scriptable plug-ins are plug-ins that have been extended to provide methods that can be called from javascript and the dom when accessed through the object or embed element. consider the following example, where a media player plug-in can be controlled with an advancetonextsong() method called inside the script element: <object id="myplugin" type="audio/wav" data="music.wav"> </object> <script type="application/javascript"> var theplugin = document.getelementbyid('myplugin'); if (theplugin && theplugin.advancetonextsong) theplugin.advancetonextsong(); else alert("plugin not installed correctly"); </script> in the past, liveconnect and later xpconnect were used to make plug-ins scriptable. see the wikipedia article for hist...

Deprecated tools - Firefox Developer Tools
canvas debugger bugzilla issue: bug 1403938 removed as of firefox 67 description canvas debugger allowed users to inspect the canvas element and see how frequently a given function is called. it was deprecated due to lack of use. we do not have dedicated documentation for the canvas debugger. alternatives spector.js is a webextension that can provide these features with 3d contexts. ...
...s debugger allowed users to inspect the canvas element and see how frequently a given function is called. it was deprecated due to lack of use. we do not have dedicated documentation for the canvas debugger. alternatives spector.js is a webextension that can provide these features with 3d contexts. ...
... canvas element and see how frequently a given function is called. it was deprecated due to lack of use. we do not have dedicated documentation for the canvas debugger. alternatives spector.js is a webextension that can provide these features with 3d contexts. web audio editor bugzilla issue: bug 1403944 removed as of firefox 67 description the web audio editor allowed you to examine an audio context constructed in the page and provided a visualization of its graph. this gave a high-level view of its operation, and enabled you to ensure that all the nodes are connected in the way you expect. it was possible to edit the audioparam properties for each node in the graph. some non-audioparam properties, like an oscillatornode's type property, were displayed and editable as well. it was de...
...tion of its graph. this gave a high-level view of its operation, and enabled you to ensure that all the nodes are connected in the way you expect. it was possible to edit the audioparam properties for each node in the graph. some non-audioparam properties, like an oscillatornode's type property, were displayed and editable as well. it was deprecated due to lack of use. more details about the web audio editor alternatives alternatives include audion and https://github.com/spite/webaudioextension web extensions. shader editor bugzilla issue: bug 1342237 removed as of firefox 67 description the shader editor allowed users to examine and edit the source of the webgl vertex and fragment shaders. it was deprecated due to low usage and maintenance costs. more details about the shader editor...

Index - Firefox Developer Tools
tools that are built into firefox. 143 validators tools, website validator, html5 validator, validator this document lists different resources for developers to validate web pages. 144 view source devtools, firefox, view page source view source lets you look at the html or xml source for the page you're viewing. ...
... validators tools, website validator, html5 validator, validator this document lists different resources for developers to validate web pages. 144 view source devtools, firefox, view page source view source lets you look at the html or xml source for the page you're viewing. ...
...site validator, html5 validator, validator this document lists different resources for developers to validate web pages. 144 view source devtools, firefox, view page source view source lets you look at the html or xml source for the page you're viewing. to activate view source: 145 web audio editor firefox, mozilla, tools, web audio api with the web audio api, developers create an audio context. ...
...pers to validate web pages. 144 view source devtools, firefox, view page source view source lets you look at the html or xml source for the page you're viewing. to activate view source: 145 web audio editor firefox, mozilla, tools, web audio api with the web audio api, developers create an audio context. ...

AnalyserNode.fftSize - Web APIs
ails in the frequency domain but fewer details in the time domain. must be a power of 2 between 252^5 and 2152^15, so one of: 32, 64, 128, 256, 512, 1024, 2048, 4096, 8192, 16384, and 32768. defaults to 2048. note: if its value is not a power of 2, or it is outside the specified range, a domexception with the name indexsizeerror is thrown. ...
...16384, and 32768. defaults to 2048. note: if its value is not a power of 2, or it is outside the specified range, a domexception with the name indexsizeerror is thrown. example the following example shows basic usage of an audiocontext to create an analysernode, then requestanimationframe and <canvas> to collect time domain data repeatedly and draw an "oscilloscope style" output of the current audio input. for more complete applied examples/information, check out our voice-change-o-matic demo (see app.js lines 128–205 for relevant code). var audioctx = new (window.audiocontext || window.webkitaudiocontext)(); var analyser = audioctx.createanalyser(); ... analyser.fftsize = 2048; var bufferlength = analyser.frequencybincount ; var dataarray = new uint8array(bufferlength); analyser.getb...
...voice-change-o-matic demo (see app.js lines 128–205 for relevant code). var audioctx = new (window.audiocontext || window.webkitaudiocontext)(); var analyser = audioctx.createanalyser(); ... ...
... working draft ...

AnalyserNode.getFloatTimeDomainData() - Web APIs
the getfloattimedomaindata() method of the analysernode interface copies the current waveform, or time-domain, data into a float32array array passed into it. if the array has fewer elements than the analysernode.fftsize, excess elements are dropped. if it has more elements than needed, excess elements are ignored. syntax var audioctx = new audiocontext(); var analyser = audioctx.createanalyser(); var dataarray = new float32array(analyser.fftsize); // float32array needs to be the same length as the fftsize analyser.getfloattimedomaindata(dataarray); // fill the float32array with data returned from getfloattimedomaindata() parameters array the float32array that the time domain data will be copied to. if the array has fewer elements than the analysernode.frequencybincount, excess elem...
...the float32array with data returned from getfloattimedomaindata() parameters array the float32array that the time domain data will be copied to. if the array has fewer elements than the analysernode.frequencybincount, excess elements are dropped. if it has more elements than needed, excess elements are ignored. return value none. ...
...er elements than the analysernode.frequencybincount, excess elements are dropped. if it has more elements than needed, excess elements are ignored. return value none. example the following example shows basic usage of an audiocontext to create an analysernode, then requestanimationframe and <canvas> to collect time domain data repeatedly and draw an "oscilloscope style" output of the current audio input. for more complete applied examples/information, check out our voice-change-o-matic-float-data demo (see the source code too). var audioctx = new (window.audiocontext || window.webkitaudiocontext)(); var analyser = audioctx.createanalyser(); ... analyser.fftsize = 1024; var bufferlength = analyser.fftsize; console.log(bufferlength); var dataarray = new float32array(bufferlength); canv...
... working draft ...

BiquadFilterNode() - Web APIs
options optional options are as follows: type: one of "lowpass", "highpass", "bandpass", "lowshelf", "highshelf", "peaking", "notch", "allpass". the meaning of the other options depends on the value of this one. ...
... options optional options are as follows: type: one of "lowpass", "highpass", "bandpass", "lowshelf", "highshelf", "peaking", "notch", "allpass". the meaning of the other options depends on the value of this one. ...
... options optional options are as follows: type: one of "lowpass", "highpass", "bandpass", "lowshelf", "highshelf", "peaking", "notch", "allpass". the meaning of the other options depends on the value of this one. ...
...ansition at the center frequency. a larger value implies a sharper transition and a larger group delay. frequency: the frequency where the center of the phase transition occurs. viewed another way, this is the frequency with maximal group delay. gain: not used. return value a new biquadfilternode object instance. specifications specification status comment web audio apithe definition of 'biquadfilternode()' in that specification. working draft initial definition. browser compatibility the compatibility table on this page is generated from structured data. if you'd like to contribute to the data, please check out https://github.com/mdn/browser-compat-data and send us a pull request. desktopmobilechromeedgefirefoxinternet exploreroperasafariandr...

BiquadFilterNode.getFrequencyResponse() - Web APIs
ameters frequencyarray a float32array containing an array of frequencies, specified in hertz, which you want to filter. magresponseoutput a float32array to receive the computed magnitudes of the freqency response for each frequency value in the frequencyarray. ...
... return value undefined exceptions invalidaccesserror the three arrays provided are not all of the same length. ...
... function calcfrequencyresponse() { biquadfilter.getfrequencyresponse(myfrequencyarray,magresponseoutput,phaseresponseoutput); for(i = 0; i <= myfrequencyarray.length-1;i++){ var listitem = document.createelement('li'); listitem.innerhtml = '<str...
... working draft ...

DelayNode() - Web APIs
the delaynode() constructor of the web audio api creates a new delaynode object with a delay-line; an audionode audio-processing module that causes a delay between the arrival of an input data, and its propagation to the output. syntax var delaynode = new delaynode(context); var delaynode = new delaynode(context, options); parameters inherits parameters from the audionodeoptions dictionary. context a reference to an audiocontext or offlineaudiocontext. options optional an object specifying the delay node options. can contain the following members: delaytime: the initial delay time for the node, in seconds. the default is 0. maxdelaytime: the maximum delay time for the node, in seconds. defaults to 1. ...
... working draft initial definition. ...
... working draft initial definition. browser compatibility the compatibility table on this page is generated from structured data. ...
...ntext. options optional an object specifying the delay node options. can contain the following members: delaytime: the initial delay time for the node, in seconds. the default is 0. maxdelaytime: the maximum delay time for the node, in seconds. defaults to 1. return value a new delaynode object instance. example const audioctx = new audiocontext(); const delaynode = new delaynode(audioctx, { delaytime: 0.5, maxdelaytime: 2, }); specifications specification status comment web audio apithe definition of 'delaynode()' in that specification. working draft initial definition. browser compatibility the compatibility table on this page is generated from structured data. if you'd like to contribute to the data, please check out https://githu...

HTMLMediaElement.readyState - Web APIs
possible values are: constant value description have_nothing 0 no information is available about the media resource. have_metadata 1 enough of the media resource has been retrieved that the metadata attributes are initialized. seeking will no longer raise an exception. have_current_data 2 data is available for the current playback position, but not enough to actually play more than one frame. ...
...a 3 data for the current playback position as well as for at least a little bit of time into the future is available (in other words, at least two frames of video, for example). have_enough_data 4 enough data is available—and the download rate is high enough—that the media can be played through to the end without interruption. ...
...east two frames of video, for example). have_enough_data 4 enough data is available—and the download rate is high enough—that the media can be played through to the end without interruption. examples this example will listen for audio data to be loaded for the element `example`. it will then check if at least the current playback position has been loaded. if it has, the audio will play. ...
...video, for example). have_enough_data 4 enough data is available—and the download rate is high enough—that the media can be played through to the end without interruption. examples this example will listen for audio data to be loaded for the element `example`. it will then check if at least the current playback position has been loaded. if it has, the audio will play. <audio id="example" preload="auto"> <source src="sound.ogg" type="audio/ogg" /> </audio> var obj = document.getelementbyid('example'); obj.addeventlistener('loadeddata', function() { if(obj.readystate >= 2) { obj.play(); } }); specifications specification status comment html living standardthe definition of 'htmlmediaelement.readystate' in that specification. ...

The HTML DOM API - Web APIs
l, as well as any supporting types and interfaces they rely upon. the functional areas included in the html dom api include: access to and control of html elements via the dom. access to and manipulation of form data. interacting with the contents of 2d images and the context of an html <canvas>, for example to draw on top of them. management of media connected to the html media elements (<audio> and <video>). dragging and dropping of content on webpages. access to the browser navigation history supporting and connective interfaces for other apis such as web components, web storage, web workers, websocket, and server-sent events. html dom concepts and usage in this article, we'll focus on the parts of the html dom that involve engaging with html elements. discussion of other areas,...
...pts surrounding the ability for the element to be contained by another element, and to contain other elements itself. of special importance is what is gained by inheriting from eventtarget: the ability to receive and handle events such as mouse clicks, play and pause events, and so forth. there are elements that share commonalities and thus have an additional intermediary type. for example, the <audio> and <video> elements both present audiovisual media. the corresponding types, htmlaudioelement and htmlvideoelement, are both based upon the common type htmlmediaelement, which in turn is based upon htmlelement and so forth. htmlmediaelement defines the methods and properties held in common between audio and video elements. these element-specific interfaces make up the majority of the html dom ...
...ual html elements, or to a small group of elements with similar functionality. in addition, the html dom api includes a few interfaces and types to support the html element interfaces. html element interfaces these interfaces represent specific html elements (or sets of related elements which have the same properties and methods associated with them). ...
... these interfaces represent objects used by the canvas api as well as the <img> element and <picture> elements. canvasgradient canvaspattern canvasrenderingcontext2d imagebitmap imagebitmaprenderingcontext imagedata offscreencanvas offscreencanvasrenderingcontext2d path2d textmetrics media interfaces the media interfaces provide html access to the contents of the media elements: <audio> and <video>. audiotrack audiotracklist mediaerror texttrack texttrackcue texttrackcuelist texttracklist timeranges trackevent videotrack videotracklist drag and drop interfaces these interfaces are used by the html_drag_and_drop_api to represent individual draggable (or dragged) items, groups of dragged or draggable items, and to handle the drag and drop process. datatransf...

IIRFilterNode - Web APIs
it lets the parameters of the filter response be specified, so that it can be tuned as needed. number of inputs 1 number of outputs 1 channel count mode "max" channel count same as on the input channel interpretation "speakers" typically, it's best to use the biquadfilternode interface to implement higher-order filters. there are several reasons why: biquad filters are typically less sensitive to numeric quirks. ...
... even-ordered iir filters can be created using biquadfilternode. however, if you need to create an odd-ordered iir filter, you'll need to use iirfilternode. you may also find this interface useful if you don't need automation, or for other reasons. once the node has been created, you can't change its coefficients. iirfilternodes have a tail-time reference; they continue to output non-silent audio with zero input. as an iir filter, the non-zero input continues forever, but this can be limited after some finite time in practice, when the output has approached zero closely enough. the actual time that takes depends on the filter coefficients provided. constructor iirfilternode() creates a new instance of an iirfilternode object. properties this interface has no properties of its own;...
...nput continues forever, but this can be limited after some finite time in practice, when the output has approached zero closely enough. the actual time that takes depends on the filter coefficients provided. constructor iirfilternode() creates a new instance of an iirfilternode object. properties this interface has no properties of its own; however, it inherits properties from its parent, audionode. methods inherits methods from its parent, audionode. it also has the following additional methods: getfrequencyresponse() uses the filter's current parameter settings to calculate the response for frequencies specified in the provided array of frequencies. examples you can find a simple iir filter demo live on codepen. also see the source code on github. it includes some differen...
...live on codepen. also see the source code on github. it includes some different coefficient values for different lowpass frequencies — you can change the value of the filternumber constant to a value between 0 and 3 to check out the different available effects. also see our using iir filters guide for a full explanation. specifications specification status comment web audio apithe definition of 'iirfilternode' in that specification. working draft initial definition ...

MediaStreamTrack.kind - Web APIs
"video": the track is a video track. ...
... "video": the track is a video track. ...
... "video": the track is a video track. ...
... "video": the track is a video track. ...

Recording a media element - Web APIs
in this article, we'll look at an example that does just that. html content <p>click the "start" button to begin video recording for a few seconds. ...
...s by calling mediastream.gettracks(), using foreach() to call mediastreamtrack.stop() on each track in the stream. ...
...the "download" button's link is then set to refer to the...
...the "download" button's link is then set to refer to the stream as well. then, in line 8, we arrange for preview.capturestream() to call preview.mozcapturestream() so that our code will work on firefox, on which the mediarecorder.capturestream() method is prefixed. ...

MediaStream Recording API - Web APIs
rom a mediastream and delivering it to you for processing. the data is delivered by a series of dataavailable events, already in the format you specify when creating the mediarecorder. you can then process the data further or write it to file as desired. overview of the recording process the process of recording a stream is simple: set up a mediastream or htmlmediaelement (in the form of an <audio> or <video> element) to serve as the source of the media data. create a mediarecorder object, specifying the source stream and any desired options (such as the container's mime type or the desired bit rates of its tracks). set mediarecorder.ondataavailable to an event handler for the dataavailable event; this will be called whenever data is available for you. once the source media is playing a...
... process of constructing the mediarecorder. to do so, you'll need to call navigator.mediadevices.enumeratedevices() to get a list of the available media devices. you can then examine that list and identify the potential input sources, and even filter the list based on desired criteria. ...
...ratedevices() is used to examine the available input devices, locate those which are audio input devices, and create <option> elements that are then added to a <select> element representing an input source picker. navigator.mediadevices.enumeratedevices() .then(function(devices) { devices.foreach(function(device) { let menu = document.getelementbyid("inputdevices"); if (device.kind == "audioinput") { let item = document.createelement("option"); item.innerhtml = device.label; item.value = device.deviceid; menu.appendchild(item); } }); }); code similar to this can be used to let the user restrict the set of devices they wish to use. for more information to learn more about using the mediastream recording api, see using the mediastream recording api, whi...
...eateelement("option"); item.innerhtml = device.label; item.value = device.deviceid; menu.appendchild(item); } }); }); code similar to this can be used to let the user restrict the set of devices they wish to use. for more information to learn more about using the mediastream recording api, see using the mediastream recording api, which shows how to use the api to record audio clips. a second article, recording a media element, describes how to receive a stream from an <audio> or <video> element and use the captured stream (in this case, recording it and saving it to a local disk). reference blobevent each time a chunk of media data is finished being recorded, it's delivered to consumers in blob form using a blobevent of type dataavailable. mediarecorder the pri...

Transcoding assets for Media Source Extensions - Web APIs
bin directory in the same place as ffmpeg. python2 — bento4 uses it. get these installed successfully before moving to the next step. sample media should be placed in the bento4 utils directory and worked here. note: the prebuilt ffmpeg does not include libfdk_aac due to licensing reasons. bento4 uses this by default, so you need to compile ffmpeg if necessary. if you don't need it, add --audio-codec=aac to the mp4-dash-encode.py command line. container and codec support as specified in section 1.1 of the mse spec: goals, mse is designed not to require support for any particular media format or codec. while this is true on paper, browser support varies for specific container/codec combinations. to check if the browser supports a particular container, you can pass a string of the mi...
... as specified in section 1.1 of the mse spec: goals, mse is designed not to require support for any particular media format or codec. while this is true on paper, browser support varies for specific container/codec combinations. to check if the browser supports a particular container, you can pass a string of the mime type to the mediasource.istypesupported method: mediasource.istypesupported('audio/mp3'); // false mediasource.istypesupported('video/mp4'); // true mediasource.istypesupported('video/mp4; codecs="avc1.4d4028, mp4a.40.2"'); // true the string is the mime type of the container, optionally followed by a list of codecs. while the mime type is fairly simple to figure out, we can get the codec string using the mp4info utility. currently, mp4 containers with h.264 video and aac aud...
...e container, optionally followed by a list of codecs. while the mime type is fairly simple to figure out, we can get the codec string using the mp4info utility. currently, mp4 containers with h.264 video and aac audio codecs have support across all modern browsers, while others don't. to convert our sample media from a quicktime mov container to an mp4 container, we can use ffmpeg. because the audio codec in the mov container is already aac and the video codec is h.264, we can instruct ffmpeg not to perform transcoding. instead, it will just copy the audio and video tracks over without performing any transcoding, which is relatively faster than having to transcode. $ ffmpeg -i trailer_1080p.mov -c:v copy -c:a copy bunny.mp4 $ ls bunny.mp4 trailer_1080p.mov checking fragmentation i...
......

Media Source API - Web APIs
we can't, for example, combine/split arraybuffers. ...
...we can't, for example, combine/split arraybuffers. ...
...h as those using dash or hls) to be built on its extensible api. creating assets that work with mse in modern browsers is a laborious process, taking significant time, computing power, and energy. the usage of external utilities to massage the content into a suitable format is required. while browser support for the various media containers with mse is spotty, usage of the h.264 video codec, aac audio codec, and mp4 container format is a common baseline. mse also provides an api for runtime detection of container and codec support. if you do not require explicit control of video quality over time, the rate at which content is fetched, or the rate at which memory is evicted, then the <video> and <source> tags may well be a simple and adequate solution. dash dynamic adaptive streaming over ht...
...then be specified as the src value of an html media element to play a media stream. htmlmediaelement.seekable when a mediasource object is played by an html media element, this property will return a timeranges object that contains the time ranges that the user is able to seek to. htmlvideoelement.getvideoplaybackquality() returns a videoplaybackquality object for the currently played video. audiotrack.sourcebuffer, videotrack.sourcebuffer, texttrack.sourcebuffer returns the sourcebuffer that created the track in question. specifications specification status comment media source extensions recommendation initial definition. ...

Media Capture and Streams API (Media Stream) - Web APIs
the media capture and streams api, often called the media streams api or simply mediastream api, is an api related to webrtc which provides support for streaming audio and video data. it provides the interfaces and methods for working with the streams and their constituent tracks, the constraints associated with data formats, the success and error callbacks when using the data asynchronously, and the events that are fired during the process. concepts and usage the api is based on the manipulation of a mediastream object representing a flux of audio- or video-related data. see an example in get the video. a mediastream consists of zero or more mediastreamtrack objects, representing various audio or video tracks. each mediastreamtrack may have one or more channels. the channel represents the ...
...oncepts and usage the api is based on the manipulation of a mediastream object representing a flux of audio- or video-related data. see an example in get the video. a mediastream consists of zero or more mediastreamtrack objects, representing various audio or video tracks. each mediastreamtrack may have one or more channels. the channel represents the smallest unit of a media stream, such as an audio signal associated with a given speaker, like left or right in a stereo audio track. mediastream objects have a single input and a single output. a mediastream object generated by getusermedia() is called local, and has as its source input one of the user's cameras or microphones. ...
... of a media stream, such as an audio signal associated with a given speaker, like left or right in a stereo audio track. mediastream objects have a single input and a single output. a mediastream object generated by getusermedia() is called local, and has as its source input one of the user's cameras or microphones. a non-local mediastream may be representing to a media element, like <video> or <audio>, a stream originating over the network, and obtained via the webrtc rtcpeerconnection api, or a stream created using the web audio api mediastreamaudiosourcenode. the output of the mediastream object is linked to a consumer. it can be a media elements, like <audio> or <video>, the webrtc rtcpeerconnection api or a web audio api mediastreamaudiosourcenode. interfaces in these reference article...
...these no longer exist, and you should update any existing code to instead use mediastreamtrack directly. events addtrack ended muted overconstrained removetrack started unmuted guides and tutorials the articles below provide additional guidance and how-to inform...

Navigator.getUserMedia() - Web APIs
if permission is denied, no compatible input devices exist, or any other error condition occurs, the error callback is executed with a mediastreamerror object describing what went wrong. ...
... if permission is denied, no compatible input devices exist, or any other error condition occurs, the error callback is executed with a mediastreamerror object describing what went wrong. ...
...traints section under the modern mediadevices.getusermedia() method, as well as the article capabilities, constraints, and settings. successcallback a function which is invoked when the request for media access is approved. the function is called with one parameter: the mediastream object that contains the media stream. your callback can then assign the stream to the desired object (such as an <audio> or <video> element), as shown in the following example: function(stream) { var video = document.queryselector('video'); video.srcobject = stream; video.onloadedmetadata = function(e) { // do something with the video here. }; } errorcallback when the call fails, the function specified in the errorcallback is invokedwith a mediastreamerror object as its sole argument; this o...
...code to cope with various browsers' prefixes. note that this is the deprecated way of doing it: see the examples section under the mediadevices.getusermedia() for modern examples. ...

OscillatorNode.OscillatorNode() - Web APIs
options optional an object whose properties specify the initial values for the oscillator node's properties. ...
... options optional an object whose properties specify the initial values for the oscillator node's properties. ...
...dionode that represents a periodic waveform, like a sine wave, optionally setting the node's properties' values to match values in a specified object. if the default values of the properties are acceptable, you can optionally use the audiocontext.createoscillator() factory method instead. syntax var oscillatornode = new oscillatornode(context, options) parameters inherits parameters from the audionodeoptions dictionary. context a reference to an audiocontext. options optional an object whose properties specify the initial values for the oscillator node's properties. any properties omitted from the object will take on the default value as documented. type the shape of the wave produced by the node. valid values are 'sine', 'square', 'sawtooth', 'triangle' and 'custom'. the defau...
... a detuning value (in cents) which will offset the frequency by the given amount. its default is 0. frequency the frequency (in hertz) of the periodic waveform. its default is 440. periodicwave an arbitrary period waveform described by a periodicwave object. return value a new oscillatornode object instance. specifications specification status comment web audio apithe definition of 'oscillatornode()' in that specification. working draft initial definition. ...

OscillatorNode.frequency - Web APIs
the frequency property of the oscillatornode interface is an a-rate audioparam representing the frequency of oscillation in hertz. syntax var oscillator = audioctx.createoscillator(); oscillator.frequency.setvalueattime(440, audioctx.currenttime); // value in hertz note: though the audioparam returned is read-only, the value it represents is not. value an a-rate audioparam. example the following example shows basic usage of an audiocontext to create an oscillator node. for an applied example, check out our violent theremin demo (see app.js for relevant code). ...
...llator.frequency.setvalueattime(440, audioctx.currenttime); // value in hertz note: though the audioparam returned is read-only, the value it represents is not. value an a-rate audioparam. example the following example shows basic usage of an audiocontext to create an oscillator node. for an applied example, check out our violent theremin demo (see app.js for relevant code). ...
...lueattime(440, audioctx.currenttime); // value in hertz note: though the audioparam returned is read-only, the value it represents is not. value an a-rate audioparam. example the following example shows basic usage of an audiocontext to create an oscillator node. for an applied example, check out our violent theremin demo (see app.js for relevant code). ...
....currenttime); // value in hertz note: though the audioparam returned is read-only, the value it represents is not. value an a-rate audioparam. example the following example shows basic usage of an audiocontext to create an oscillator node. for an applied example, check out our violent theremin demo (see app.js for relevant code). // create web audio api context var audioctx = new (window.audiocontext || window.webkitaudiocontext)(); // create oscillator node var oscillator = audioctx.createoscillator(); oscillator.type = 'square'; oscillator.frequency.setvalueattime(440, audioctx.currenttime); // value in hertz oscillator.start(); specifications specification status comment web audio apithe definition of 'frequency' in that specification. working draft ...

PannerNode.coneInnerAngle - Web APIs
the coneinnerangle property of the pannernode interface is a double value describing the angle, in degrees, of a cone inside of which there will be no volume reduction. the coneinnerangle property's default value is 360, suitable for a non-directional source. syntax var audioctx = new audiocontext(); var panner = audioctx.createpanner(); panner.coneinnerangle = 360; value a double. example in this example, we'll demonstrate how changing the orientation parameters of a pannernode in combination with coneinnerangle and coneouterangle affects volume. to help us visualise how the orientation vector affects, we can use the right-hand rule: first, let's start by writing a utility function to figure out our orientation vector. the x and z components are always at a 90° to each other, so ...
... offset the angle so 0 points towards the listener const radians = (degrees - 90) * (math.pi / 180); // using cosine and sine here ensures the output values are always normalised // i.e. ...
...= (degrees - 90) * (math.pi / 180); // using cosine and sine here ensures the output values are always normalised // i.e. ...
... working draft ...

PannerNode.coneOuterAngle - Web APIs
the coneouterangle property of the pannernode interface is a double value describing the angle, in degrees, of a cone outside of which the volume will be reduced by a constant value, defined by the coneoutergain property. the coneouterangle property's default value is 0. syntax var audioctx = new audiocontext(); var panner = audioctx.createpanner(); panner.coneouterangle = 0; value a double. example in this example, we'll demonstrate how changing the orientation parameters of a pannernode in combination with coneinnerangle and coneouterangle affects volume. to help us visualise how the orientation vector affects, we can use the right-hand rule: first, let's start by writing a utility function to figure out our orientation vector. the x and z components are always at a 90° to each ...
... offset the angle so 0 points towards the listener const radians = (degrees - 90) * (math.pi / 180); // using cosine and sine here ensures the output values are always normalised // i.e. ...
...= (degrees - 90) * (math.pi / 180); // using cosine and sine here ensures the output values are always normalised // i.e. ...
... working draft ...

PannerNode.positionX - Web APIs
the positionx property of the pannernode interface specifies the x coordinate of the audio source's position in 3d cartesian coordinates, corresponding to the horizontal axis (left-right). the complete vector is defined by the position of the audio source, given as (positionx, positiony, positionz), and the orientation of the audio source (that is, the direction in which it's facing), given as (orientationx, orientationy, orientationz). depending on the directionality of the sound (as specified using the attributes coneinnerangle, coneouterangle, and codeoutergain), the orientation of the sound may alter the perceived volume of the sound as it's being played. ...
...en as (orientationx, orientationy, orientationz). depending on the directionality of the sound (as specified using the attributes coneinnerangle, coneouterangle, and codeoutergain), the orientation of the sound may alter the perceived volume of the sound as it's being played. if the sound is pointing toward the listener, it will be louder than if the sound is pointed away from the listener. the audioparam contained by this property is read only; however, you can still change the value of the parameter by assigning a new value to its audioparam.value property. syntax var positionx = pannernode.positionx; pannernode.positionx.value = newpositionx; value an audioparam whose value is the x coordinate of the audio source's position, in 3d cartesian coordinates. the default value is 0. exam...
...itionx = pannernode.positionx; pannernode.positionx.value = newpositionx; value an audioparam whose value is the x coordinate of the audio source's position, in 3d cartesian coordinates. the default value is 0. example the following example starts an oscillator, and pans it to the left after 1 second, to the right after 2 seconds, and back to the center after 3 seconds. ...
... working draft ...

PannerNode.positionY - Web APIs
the positiony property of the pannernode interface specifies the y coordinate of the audio source's position in 3d cartesian coordinates, corresponding to the vertical axis (top-bottom). the complete vector is defined by the position of the audio source, given as (positionx, positiony, positionz), and the orientation of the audio source (that is, the direction in which it's facing), given as (orientationx, orientationy, orientationz). depending on the directionality of the sound (as specified using the attributes coneinnerangle, coneouterangle, and codeoutergain), the orientation of the sound may alter the perceived volume of the sound as it's being played. ...
...en as (orientationx, orientationy, orientationz). depending on the directionality of the sound (as specified using the attributes coneinnerangle, coneouterangle, and codeoutergain), the orientation of the sound may alter the perceived volume of the sound as it's being played. if the sound is pointing toward the listener, it will be louder than if the sound is pointed away from the listener. the audioparam contained by this property is read only; however, you can still change the value of the parameter by assigning a new value to its audioparam.value property. syntax var positiony = pannernode.positiony; pannernode.positiony.value = newpositiony; value an audioparam whose value is the y coordinate of the audio source's position, in 3d cartesian coordinates. example the following examp...
...e value is the y coordinate of the audio source's position, in 3d cartesian coordinates. example the following example starts an oscillator and pans it above the listener after 1 second, below the listener after 2 seconds, and back to the center after 3 seconds. note that in this case, the change will mainly affect the timbre of the oscillator, as it's a simple mono wave. ...
... working draft ...

WaveShaperNode.oversample - Web APIs
once applied, the number of samples is reduced to its initial numbers. this leads to better results by avoiding some aliasing, but comes at the expense of a lower precision shaping curve. the possible oversample values are: value effect 'none' do not perform any oversampling. ...
...rform any oversampling. '2x' double the amount of samples before applying the shaping curve. '4x' multiply by 4 the amount of samples before applying the shaping curve. syntax distortion.oversample = enumeratedvalue; values distortion is a waveshapernode. enumeratedvalue is one of 'none', '2x', or '4x'. ...
...usage of an audiocontext to create a wave shaper node. for applied examples/information, check out our voice-change-o-matic demo (see app.js for relevant code). note: sigmoid functions are commonly used for distortion curves because of their natural properties. their s-shape, for instance, helps create a smoother sounding result. we found the below distortion curve code on stack overflow. var audioctx = new (window.audiocontext || window.webkitaudiocontext)(); var distortion = audioctx.createwaveshaper(); ... function makedistortioncurve(amount) { var k = typeof amount === 'number' ? amount : 50, n_samples = 44100, curve = new float32array(n_samples), deg = math.pi / 180, i = 0, x; for ( ; i < n_samples; ++i ) { x = i * 2 / n_samples - 1; curve[i] = ( 3 +...
... working draft ...

WebRTC Statistics API - Web APIs
local-candidate statistics about a local ice candidate associated with the connection's rtcicetransports. rtcicecandidatestats rtcstats media-source statistics about the media produced by the mediastreamtrack attached to an rtp sender. the dictionary this key maps to depends on the track's kind. rtcaudiosourcestats or rtcvideosourcestats rtcmediasourcestats rtcstats rtcaudiosourcestats or rtcvideosourcestats outbound-rtp statistics describing the state of one of the outbound data streams on this connection. rtcoutboundrtpstreamstats rtcsentrtpstreamstats rtcrtpstreams...
... peer-connection statistics describing the state of the rtcpeerconnection. rtcpeerconnectionstats rtcstats receiver statistics related to a specific rtcrtpreceiver and the media associated with that receiver. the specific type of object representing the statistics depends on the media kind. rtcaudioreceiverstats or rtcvideoreceiverstats rtcaudiohandlerstats or rtcvideohandlerstats rtcmediahandlerstats rtcstats remote-candidate statistics about a remote ice candidate associated with the connection's rtcicetransports. rtcicecandidatestats r...
... sctp-transport statistics about an rtcsctptransport. rtcsctptransportstats rtcstats sender statistics related to a specific rtcrtpsender and the media associated with that sender. the type of object representing this statistic depends on the kind of the media. rtcaudiosenderstats or rtcvideosenderstats rtcaudiohandlerstats or rtcvideohandlerstats rtcmediahandlerstats rtcstats stream statistics about a particular media mediastream. this has been obsoleted since the transition to webrtc being track-based rather than stream-based. ...
... mediastreamtrack's attachment to one of the connection's senders or receivers. the referenced object's type depends on the track type. these statistics have all been moved to media-source, sender, receiver, outbound-rtp, and inbound-rtp, and this statistic category type is thus obsolete and shouldn't be used anymore. rtcsendervideotrackattachmentstats or rtcsenderaudiotrackattachmentstats or rtcreceivervideotrackattachmentstats or rtcreceiveraudiotrackattachmentstats rtcmediahandlerstats rtcstats transceiver statistics related to a specific rtcrtptransceiver. rtcrtptransceiverstats rtcstats transport statistics about...

Setting up adaptive streaming media sources - Developer guides
giving its earliest presentation time and other meta-data and can often make up a large portion of the mpd file. ondemand profile this profile will allow switching between streams 'on demand' - that is to say that you only need provide a set of contiguous files and specify the bandwidth for each one and the appropriate file will be chosen automatically. here's a simple example that provides an audio track representation and four separate video representations. <?xml version="1.0" encoding="utf-8"?> <mpd xmlns:xsi="http://www.w3.org/2001/xmlschema-instance" xmlns="urn:mpeg:dash:schema:mpd:2011" xsi:schemalocation="urn:mpeg:dash:schema:mpd:2011 dash-mpd.xsd" type="static" mediapresentationduration="pt654s" minbuffertime="pt2s" profiles="urn:mpeg:dash:profile:isoff-on-demand:2011">...
....0" encoding="utf-8"?> <mpd xmlns:xsi="http://www.w3.org/2001/xmlschema-instance" xmlns="urn:mpeg:dash:schema:mpd:2011" xsi:schemalocation="urn:mpeg:dash:schema:mpd:2011 dash-mpd.xsd" type="static" mediapresentationduration="pt654s" minbuffertime="pt2s" profiles="urn:mpeg:dash:profile:isoff-on-demand:2011"> <baseurl>http://example.com/ondemand/</baseurl> <period> <!-- english audio --> <adaptationset mimetype="audio/mp4" codecs="mp4a.40.5" lang="en" subsegmentalignment="true" subsegmentstartswithsap="1"> <representation id="1" bandwidth="64000"> <baseurl>elephantsdream_aac48k_064.mp4.dash</baseurl> </representation> </adaptationset> <!-- video --> <adaptationset mimetype="video/mp4" codecs="avc1.42401e" subsegmentalignment="true" subsegme...
...5972"/> </segmentlist> </representation> ... </adaptationset> </period> </mpd> the mpd file tells the browser where the various pieces of media are located, it also includes meta data such as mimetype and codecs and there are other details such as byte-ranges in there too. generally these files will be generated for you. note: you can also split out your audio and video streams into separate files, which can then be prioritised and served separately depending on bandwidth. once you have generated your mpd file you can reference as expected it from within the <video> element: <video src="my.mpd" type="application/dash+xml"></video> it might be wise to provide a fallback: <video> <source src="my.mpd" type="application/dash+xml"> <!-- fallback --...
...es on dash.js and browser support for media source extensions, see the latest dash.js reference player. hls encoding http live streaming (hls) is an http-based media streaming protocol implemented by apple. it's incorporated into ios and osx platforms and works well on mobile and desktop safari and most android devices with some caveats. media is usually encoded as mpeg-4 (h.264 video and aac audio) and packaged into an mpeg-2 transport stream, which is then broken into segments and saved as one or more .ts media files. apple provides tools to convert media files to the appropriate format. hls encoding tools ther are a number of useful tools available for hls encoding for hls live stream encoding adobe provide a media encoder for mac. the stream segmenter — provided by apple for mac...

Content categories - Developer guides
r the behavior of the rest of the document, set up links to other documents, or convey other out of band information. elements belonging to this category are <base>, <command>, <link>, <meta>, <noscript>, <script>, <style> and <title>. flow content elements belonging to the flow content category typically contain text or embedded content. ...
...ontent itself. note: the <hgroup> element was removed from the w3c html specification prior to html 5 being finalized, but is still part of the whatwg specification and is at least partially supported by most browsers. phrasing content phrasing content defines the text and the mark-up it contains. runs of phrasing content make up paragraphs. ...
...ains only phrasing content <ins>, if it contains only phrasing content <link>, if the itemprop attribute is present <map>, if it contains only phrasing content <meta>, if the itemprop attribute is present embedded content embedded content imports another resource or inserts content from another mark-up language or namespace into the document. elements that belong to this category include: <audio>, <canvas>, <embed>, <iframe>, <img>, <math>, <object>, <picture>, <svg>, <video>. interactive content interactive content includes elements that are specifically designed for user interaction. elements that belong to this category include: <a>, <button>, <details>, <embed>, <iframe>, <keygen>, <label>, <select>, and <textarea>. some elements belong to this category only under specific condit...
...>, <embed>, <iframe>, <img>, <math>, <object>, <picture>, <svg>, <video>. interactive content interactive content includes elements that are specifically designed for user interaction. elements that belong to this category include: <a>, <button>, <details>, <embed>, <iframe>, <keygen>, <label>, <select>, and <textarea>. ...

DASH Adaptive Streaming for HTML 5 Video - HTML: Hypertext Markup Language
media source extensions api which will allow support for dash via javascript libraries such as dash.js. see bug 778617 for details. using dash - server side first you'll need to convert your webm video to a dash manifest with the accompanying video files in various bit rates. ...
...778617 for details. using dash - server side first you'll need to convert your webm video to a dash manifest with the accompanying video files in various bit rates. to start with you'll only need the ffpmeg program from ffmpeg.org, with libvpx and libvorbis support for webm video and audio, at least version 2.5 (probably; this was tested ith 3.2.5). 1. use your existing webm file to create one audio file and multiple video files. for example: the file in.video can be any container with at least one audio and one video stream that can be decoded by ffmpeg, create the audio using: ffmpeg -i in.video -vn -acodec libvorbis -ab 128k -dash 1 my_audio.webm create each video variant. ffmpeg -i in.video -c:v libvpx-vp9 -keyint_min 150 -g 150 -tile-columns 4 -frame-parallel 1 -f webm -dash 1 ...
...60 -b:v 1000k -dash 1 video_640x360_1000k.webm \ -an -vf scale=1280:720 -b:v 1500k -dash 1 video_1280x720_1500k.webm 2. create the manifest file: ffmpeg \ -f webm_dash_manifest -i video_160x90_250k.webm \ -f webm_dash_manifest -i video_320x180_500k.webm \ -f webm_dash_manifest -i video_640x360_750k.webm \ -f webm_dash_manifest -i video_1280x720_1500k.webm \ -f webm_dash_manifest -i my_audio.webm \ -c copy \ -map 0 -map 1 -map 2 -map 3 -map 4 \ -f webm_dash_manifest \ -adaptation_sets "id=0,streams=0,1,2,3 id=1,streams=4" \ my_video_manifest.mpd the -map arguments correspond to the input files in the sequence they are given; you should have one for each file. the -adaptation_sets argument assigns them into adaptation sets; for example, this creates one set (0) that contai...
...on_sets "id=0,streams=0,1,2,3 id=1,streams=4" \ my_video_manifest.mpd the -map arguments correspond to the input files in the sequence they are given; you should have one for each file. the -adaptation_sets argument assigns them into adaptation sets; for example, this creates one set (0) that contains the streams 0, 1, 2 and 3 (the videos), and another set (1) that contains only stream 4, the audio stream. put the manifest and the associated video files on your web server or cdn. dash works via http, so as long as your http server supports byte range requests, and it's set up to serve .mpd files with mimetype="application/dash+xml", then you're all set. using dash - client side you'll want to modify your web page to point to the dash manifest first, instead of directly to a particular vi...

<track>: The Embed Text Track element - HTML: Hypertext Markup Language
the html <track> element is used as a child of the media elements, <audio> and <video>. it lets you specify timed text tracks (or time-based data), for example to automatically handle subtitles. the tracks are formatted in webvtt format (.vtt files) — web video text tracks. the source for this interactive example is stored in a github repository. if you'd like to contribute to the interactive examples project, please clone https://github.com/mdn/interactive-examples and send us a pull request. content categories none permitted content none, it is an empty element. tag omission as it is a void element, the start tag must be present and the end tag must not be present. permitted parents a media element, <audio> or <video>. i...
...cannot be understood by the viewer. for example dialogue or text that is not english in an english language film. subtitles may contain additional content, usually extra background information. for example the text at the beginning of the star wars films, or the date, time, and location of a scene. captions closed captions provide a transcription and possibly a translation of audio. it may include important non-verbal information such as music cues or sound effects. it may indicate the cue's source (e.g. music, text, character). suitable for users who are deaf or when the sound is muted. descriptions textual description of the video content. suitable for users who are blind or where the video cannot be seen. chapters chapter tit...
...navigating the media resource. metadata tracks used by scripts. not visible to the user. label a user-readable title of the text track which is used by the browser when listing available text tracks. src address of the track (.vtt file). must be a valid url. this attribute must be specified and its url value must have the same origin as the document — unless the <audio> or <video> parent element of the track element has a crossorigin attribute. srclang language of the track text data. it must be a valid bcp 47 language tag. if the kind attribute is set to subtitles, then srclang must be defined. usage notes track data types the type of data that track adds to the media is set in the kind attribute, which can take values of subtitles, captions, description...
...t cannot have more than one track with the same kind, srclang, and label. detecting cue changes the underlying texttrack, indicated by the track property, receives a cuechange event every time the currently-presented cue is changed. this happens even if the track isn't associated with a media element. if the track is associated with a media element, using the <track> element as a child of the <audio> or <video> element, the cuechange event is also sent to the htmltrackelement. let texttrackelem = document.getelementbyid("texttrack"); texttrackelem.addeventlistener("cuechange", (event) => { let cues = event.target.track.activecues; }); in addition, you can use the oncuechange event handler: let texttrackelem = document.getelementbyid("texttrack"); texttrackelem.oncuechange = (event) =...

Index of archived content - Archive of obsolete content
on gecko coding help wanted http class overview hacking wiki help viewer creating a help content pack helper apps (and a bit of save as) hidden prefs how to write and land nanojit patches io guide/directory keys introducing the audio api extension isp data java in firefox extensions javascript os.shared javascript crypto crmf request object generatecrmfrequest() importusercertificates popchallengeresponse jetpack basics co...
...owsers on windows mcd, mission control desktop, aka autoconfig monitoring wifi access points no proxy for configuration notes on html reflow same-origin policy for file: uris source navigator source code directories overview using xml data islands in mozilla using content preferences visualizing an audio spectrum working with bfcache cert_override.txt mozilla release faq newsgroup summaries format mozilla.dev.apps.firefox-2006-09-29 mozilla.dev.apps.firefox-2006-10-06 mozilla-dev-accessibility 2006-10-06 2006-11-10 2006-11-2...
... references summary of changes using the w3c dom using workers in extensions web standards choosing standards compliance over proprietary practices community correctly using titles with external stylesheets describing microformats in javascript displaying a graphic with audio samples fixing incorrectly sized list item markers fixing table inheritance in quirks mode issues arising from arbitrary-element hover mozilla's doctype sniffing parsing microformats in javascript popup window controls rdf in fifty words or less rdf in mozilla faq styling abbreviations and acronyms ...

Assessment: Accessibility troubleshooting - Learn web development
tting it in a proper html5 semantic element. which one should it be updated to? make the update. note: you'll need to update the css rule selectors that style the tags to their proper equivalents for the semantic headings. once you add paragraph elements, you'll notice the styling looking better. the images the images are currently inaccessible to screenreader users. ...
...tml5 semantic element. which one should it be updated to? make the update. note: you'll need to update the css rule selectors that style the tags to their proper equivalents for the semantic headings. once you add paragraph elements, you'll notice the styling looking better. the images the images are currently inaccessible to screenreader users. ...
...the tags to their proper equivalents for the semantic headings. once you add paragraph elements, you'll notice the styling looking better. the images the images are currently inaccessible to screenreader users. can you fix this? the audio player the <audio> player isn't accessible to hearing impaired (deaf) people — can you add some kind of accessible alternative for these users? the <audio> player isn't accessible to those using older browsers that don't support html5 audio. how can you allow them to still access the audio? the forms the <input> element in the search form at the top could do with a label, but we don't want to add a visible text label that would potentially spoil the design and isn't really needed by sighted users. how can you add a label that is only accessibl...

How can we design for all types of users? - Learn web development
le the div either to set its width (using the width property) or its maximum width so that it never gets too large (using the max-width property). if you want an elastic/responsive website, and you don't know what the browser's default width is, you can use the max-width property to allow up to 70 characters per line and no more: div.container { max-width:70em; } alternative content for images, audio, and video websites often include stuff besides plain text. images images can be either decorative or informative, but there's no guarantee that your users can see them. for example, visually impaired users rely on a screen reader, which can only handle text. your readers may be using a very strict intranet that blocks images originating from a cdn. your readers may have disabled images t...
...ie chart with a table providing the same data), or resort to a longdesc attribute. this attribute's value is a url pointing towards a resource explicitly describing in detail the image's content. note: the use and even the existence of longdesc has been debated for quite some time. please refer to the w3c's image description extension (longdesc) for the full explanation and thorough examples. audio/video you must also provide alternatives to multimedia content. subtitling/close-captioning you should include captions in your video to cater to visitors who can't hear the audio. some users have hearing challenges, lack functioning speakers, or work in a noisy environment (like on the train). transcript subtitles only work if somebody watches the video. many users don't have time, or lac...
... image compression some users may choose to display images, but still have limited bandwidth available, especially in developing countries and on mobile devices. if you want a successful website, please compress your images. there are various tools to help you, either online or local: installed software. ...

How much does it cost to do something on the Web? - Learn web development
s to standard file formats, but each editor saves ongoing projects in its own specialized format. most of the images on the internet are copyrighted, so it is better to check the license of the file before you use it. sites like pixabay provide images under cc0 license, so you can use, edit and publish them even with modification for commercial use. media editors if you want to include video or audio into your website, you can either embed online services (for example youtube, vimeo, or dailymotion), or include your own videos (see below for bandwidth costs). for audio files, you can find free software (audacity, wavosaur), or paying up to a few hundred dollars (sony sound forge, adobe audition). likewise, video-editing software can be free (pitivi, openshot for linux, imovie for mac), less ...
.... web access computer / modem you need a computer. costs can vary tremendously, depending on your budget, and where you live. to publish a barebones website, you only need a basic computer capable of launching an editor, and a web browser, so the entry level can be quite low. of course, you'll need a more serious computer if you want to produce complicated designs, touch up photos, or produce audio and video files. you need to upload content to a remote server (see hosting below), so you need a modem. your isp can rent internet connectivity to you for a few dollars per month, though your budget might vary, depending on your location. isp access make sure that you have sufficient bandwidth: low-bandwidth access may be adequate to support a 'simple' website: reasonably-sized images, tex...
...w-bandwidth access may be adequate to support a 'simple' website: reasonably-sized images, texts, some css and javascript. that will likely cost you a few dozen dollars, including the rent for the modem. on the other hand, you'll need a high-bandwidth connection, such as dsl, cable, or fiber access, if you want a more advanced website with hundreds of files, or if you want to deliver heavy video/audio files directly from your web server. it could cost the same as low-bandwidth access, upwards to several hundred dollars per month for more professional needs. hosting understanding bandwidth hosting providers charge you according to how much bandwidth your website consumes. this depends on how many people, and web crawling robots, access your content during a given time, and how much server s...

What is a URL? - Learn web development
andling parameters is by asking the web server owner. #somewhereinthedocument is an anchor to another part of the resource itself. an anchor represents a sort of "bookmark" inside the resource, giving the browser the directions to show the content located at that "bookmarked" spot. on an html document, for example, the browser will scroll to the point where the anchor is defined; on a video or audio document, the browser will try to go to the time the anchor represents. it is worth noting that the part after the #, also known as the fragment identifier, is never sent to the server with the request. note: there are some extra parts and some extra rules regarding urls, but they are not relevant for regular users or web developers. don't worry about this, you don't need to know them to build ...
...using ftp, for example, is not secure and is no longer supported ...
...nk a document with its related resources through various elements such as <link> or <script>; to display media such as images (with the <img> element), videos (with the <video> element), sounds and music (with the <audio> element), etc.; to display other html documents with the <iframe> element. note: when specifying urls to load resources as part of a page (such as when using the <script>, <audio>, <img>, <video>, and the like), you should generally only use http and https urls, with few exceptions (one notable one being data:; see data urls). using ftp, for example, is not secure and is no longer supported by modern browsers. other technologies, such as css or javascript, use urls extensively, and these are really the heart of the web. absolute urls vs relative urls what we saw above...

Images in HTML - Learn web development
appear when the image is gone. try turning images off in your browser and see how it looks. a figure doesn't have to be an image. it is an independent unit of content that: expresses your meaning in a compact, easy-to-grasp way. could go in several places in the page's linear flow. provides essential information supporting the main text. a figure could be several images, a code snippet, audio, video, equations, a table, or something else. active learning: creating a figure in this active learning section, we'd like you to take the finished code from the previous active learning section, and turn it into a figure: wrap it in a <figure> element. copy the text out of the title attribute, remove the title attribute, and put the text inside a <figcaption> element below the image. i...
... you've reached the end of this article, but can you remember the most important information? you can find some further tests to verify that you've retained this information before you move on — see test your skills: html images. summary that's all for now. ...
...urther tests to verify that you've retained this information before you move on — see test your skills: html images. summary that's all for now. we have covered images and captions in detail. ...

From object to iframe — other embedding technologies - Learn web development
at this point we'd like to take somewhat of a sideways step, looking at some elements that allow you to embed a wide variety of content types into your webpages: the <iframe>, <embed> and <object> elements. <iframe>s are for embedding other web pages, and the other two allow you to embed pdfs, svg, and even flash — a technology that is on the way out, but which you'll still see semi-regularly. ...
...technologies. to start with, you are unlikely to use embedding for much beyond including third-party content like maps and videos on your pages. as you become more experienced, however, you are likely to start finding more uses for them. there are many other technologies that involve embedding external content besides the ones we discussed here. ...
... besides the ones we discussed here. we saw some in earlier articles, such as <video>, <audio>, and <img>, but there are others to discover, such as <canvas> for javascript-generated 2d and 3d graphics, and <svg> for embedding vector graphics. we'll look at svg in the next article of the module. ...

Graceful asynchronous programming with Promises - Learn web development
ices enabled, and directly returning the mediastream for the stream created from the selected sources, getusermedia() returns a promise which is resolved with the mediastream once it's available. ...
...rconnection.addtrack(track, chatstream)); setstatusmessage("connected"); }).catch(err => { setstatusmessage("failed to connect"); }); } this function starts by using a function called setstatusmessage() to update a status display with the message "calling...", indicating that a call is being attempted. it then calls getusermedia(), asking for a stream that has both video and audio tracks, then once that's been obtained, sets up a video element to show the stream coming from the camera as a "self view," then takes each of the stream's tracks and adds them to the webrtc rtcpeerconnection representing a connection to another user. after that, the status display is updated to say "connected". if getusermedia() fails, the catch block runs. this uses setstatusmessage() to updat...
...l be a problem is in opera mini and ie11 and earlier versions. we didn't touch on all promise features in this article, just the most interesting and useful ones. as you start to learn more about promises, you'll come across further features and techniques. most modern web apis are promise-based, so you'll need to understand promises to get the most out of them. among those apis are webrtc, web audio api, media capture and streams, and many more. promises will be more and more important as time goes on, so learning to use and understand them is an important step in learning modern javascript. ...

Handling common accessibility problems - Learn web development
diately think of making sure websites/apps are usable by people with disabilities, for example: visually impaired people using screen readers or magnification/zoom to access text people with motor function impairments using the keyboard (or other non-mouse features) to activate website functionality. people with hearing impairments relying on captions/subtitles or other text alternatives for audio/video content. however, it is wrong to say that accessibility is just about disabilities. really, the aim of accessibility is to make your websites/apps usable by as many people in as many contexts as possible, not just those users using high-powered desktop computers. some examples might include: users on mobile devices. users on alternative browsing devices such as tvs, watches, etc. use...
...in relevant content. this should contain a description of the image that successfully conveys its meaning and content on the page, to be picked up by a screenreader and read out to the user. note: for more information, read text alternatives. missing alt text can be tested for in a number of ways, for example using accessibility auditing tools. ...
... element relationships and context there are certain features and best practices in html designed to provide context and relationships between elements where none otherwise exists. ...

Handling common HTML and CSS problems - Learn web development
or id that you are already using for a different purpose. to ensure this doesn't happen, you need to research the tools you are using first and design your code around them. it is also worth "namespacing" css, e.g. if you have a widget, make sure it has a distinct class, and then start the selectors that select elements inside the widget with this class, so conflicts are less likely. for example .audio-player ul a. validation for html, validation involves making sure all your tags are properly closed and nested, you are using a doctype, and you are using tags for their correct purpose. a good strategy is to validate your code regularly. one service that can do this is the w3c markup validation service, which allows you to point to your code, and returns a list of errors: css has a similar...
...hen you need to support old browsers (such as old ie versions) or you are using features that are implemented using css prefixes. in general, most core html and css functionality (such as basic html elements, css basic colors and text styling) works across most browsers you'll want to support; more problems are uncovered when you start wanting to use newer features such as flexbox, or html5 video/audio, or even more nascent, css grids or -webkit-background-clip: text. once you've identified a list of potential problem technologies you will be using, it is a good idea to research what browsers they are supported in, and what related techniques are useful. see finding help below. html fallback behaviour some problems can be solved by just taking advantage of the natural way in which html/css w...
...ting display: block; on all of the new semantic elements (such as <article>, <aside>, etc.), but only in old versions of ie that don't recognise them (so, ie 8 and lower). this way new browsers can just use the code as normal, but older ie versions will be able to style these elements too. note: see ie conditional comments for the best way to do this. more complex elements like html <video>, <audio>, and <canvas> (and other features besides) have natural mechanisms for fallbacks to be added, which work on the same principle as described above. you can add fallback content in between the opening and closing tags, and non-supporting browsers will effectively ignore the outer element and run the nested content. for example: <video id="video" controls preload="metadata" poster="img/poster.jpg...

Index
nterface is used to specify options for geolocation operations. 494 nsidomglobalpropertyinitializer needscontent, reference, référence(2), xpcom, xpcom interface reference an xpcom component implementing this interface can be exposed to web content as a property on the window. see custom window properties are a snap in firefox blog post for details. 495 nsidomhtmlaudioelement interfaces, interfaces:scriptable, xpcom interface reference the nsidomhtmlaudioelement interface is used to implement the html5 <audio> element. 496 nsidomhtmlformelement xpcom, xpcom interface reference the nsidomhtmlformelement interface implements the dom htmlformelement interface. see that page for details. 497 nsidomhtmlmediaelement firef...
...page for details. 497 nsidomhtmlmediaelement firefox 3.5, gecko 1.9.1, interfaces, interfaces:scriptable, xpcom, xpcom interface reference no summary! 498 nsidomhtmlsourceelement dom, html, html5, interfaces, interfaces:scriptable, media, xpcom, xpcom interface reference the nsidomhtmlsourceelement interface is the dom interface to the source child of the audio and video media elements in html. 499 nsidomhtmltimeranges interfaces, media, xpcom interface reference the nsidomhtmltimeranges interface represents a set of time ranges in media; it's primarily used by the nsidomhtmlmediaelement interface, and implements the dom timeranges interface. 500 nsidomjswindow interfaces, xpcom, xpcom api reference, xpcom interface r...
... running in a window, simply use new domparser(). when you create a domparser from a privileged script, you can pass parameters to the constructor, more on that below. 512 nsidomprogressevent interfaces, interfaces:scriptable, reference, xmlhttprequest, xpcom interface reference, nsidomprogressevent, progress the nsidomprogressevent is used in the media elements (<video> and <audio>) to inform interested code of the progress of the media download. this implementation is a placeholder until the specification is complete, and is compatible with the webkit progressevent. 513 nsidomserializer dom, interfaces, interfaces:scriptable implemented by: @mozilla.org/xmlextras/xmlserializer;1. to create an instance, use: 514 nsidomsimplegestureevent ...