Copyright © 2012W3C® (MIT,ERCIM,Keio), All Rights Reserved. W3Cliability,trademark anddocument use rules apply.
A number of existing or proposed features for the Web platform deal with continuous real-time media:
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This specification documents a proposal for an audio processing and synthesis API for use in client-side user agents (e.g. a browser). This document was produced by members of theAudio Working Group. It was initially published alongside an alternative proposal, the Web Audio API. On May 9th, 2012, the Audio Working Groupresolved to re-publish this proposal as a Working Group Note, with a view to use this document as input for the Web Audio API draft specification.
If you wish to make comments regarding this document, please send them topublic-audio@w3.org including the prefix'[MSP NOTE comment]'
in the subject line. The archives of the public-audio mailing list are publicly available.
Web content and browser developers especially are encouraged to review this draft, and to experiment with the API and provide feedback.
Publication as a Working Group Note does not imply endorsement by the W3C Membership. This is a draft document and may be updated, replaced or obsoleted by other documents at any time. It is inappropriate to cite this document as other than work in progress.
This document was produced by a group operating under the5 February 2004W3C Patent Policy.W3C maintains apublic list of any patent disclosures made in connection with the deliverables of the group; that page also includes instructions for disclosing a patent. An individual who has actual knowledge of a patent which the individual believes containsEssential Claim(s) must disclose the information in accordance withsection 6 of theW3C Patent Policy.
The ideas here build onIan Hickson's proposal for HTML Streams, adding features partly inspired by the Mozilla audio API andthe Chrome audio API. Unlike previous audio API proposals, the API presented here integrates with proposed API for media capture from local devices, integrates with proposed API for peer-to-peer media streaming, handles audio and video in a unified framework, incorporates Worker-based Javascript audio processing, and specifies synchronization across multiple media sources and effects. The API presented here does not include a library of "native" effects; those should be added as new "named effects" (described below), perhaps as a "level 2" spec.
The work here is nascent. Until a prototype implementation exists, this proposal is likely to be incomplete and possibly not even implementable.
These are concrete usage scenarios that have helped guide the design of the API. They are higher-level than use-cases.
The description of MediaStreams here extends and must remain compatible with
HTML MediaStreams.Each MediaStream DOM object has an underlying media stream. The underlying media streams form a graph;
some streams (represented by ProcessedMediaStream DOM objects) can take other streams as inputs and compute an output stream.To avoid interruptions due to script execution, script execution can overlap with media stream processing;
media streams continue to play and change state during script execution. However, to simplify the DOM programming model, we limit the interaction of MediaStream DOM objects with their underlying media streams. Specifically: stream.inputs[0].volume = 0; if (needToPlay()) { stream.inputs[0].volume = 1.0; }
Specify exactly which attributes (and methods) are subject to this regime, possibly extending to attributes and methods already defined in HTML for media elements etc.
In this spec, references toMediaStream
s andMediaInput
s refer to the DOM-visible state, and references tomedia streams andinput ports refer to the underlying real-time media stream graph.
A stream is an abstraction of a time-varying video and/or audio signal. At a given point in time, a media stream can beblocked, that is, not playing for some reason. All non-blocked streams play at the same constant rate --- real time. Streams cannot be seeked or played at a rate other than real time. For convenience we define a stream's "current time" as the duration it has played since it was created, but (unless blocking is involved) streams do not have independent timelines; they are synchronized.
At the implementation level, and when processing media data with Workers, we assume that each stream has a buffered window of media data available, containing the sample it is currently playing (or that it will play the next time it unblocks). This buffer defines the stream's contents into the future.
A stream can be in thefinished state. A finished stream is always blocked and can
never leave the finished state --- it will never produce any more content.A stream that has no consumers must block. Stream consumers defined in this specification are media elements andProcessedMediaStream
s (see below). This avoids situations where streams that aren't connected
ProcessMediaEvent
is being sent). A muted audio element can be used as a dummy sink if necessary.We do not allow streams to have independent timelines (e.g. no adjustable playback
rate or seeking within an arbitrary stream), because that can lead to a single stream being consumed at multiple different "current times" simultaneously, which requires either unbounded buffering or multiple internal decoders and buffers for a single stream. It seems simpler and more predictable for performance to require authors to create multiple streams (if necessary) and change the playback rate in the original stream sources to handle such situations.For example, consider this hard case:
Authors can avoid this by explicitly splitting streams that may need to progress at
different rates --- in the above case, by using two separate media elements each loading http://fast. The HTML spec encourages implementations to share cached media data between media elements loading the same URI.A media stream contains video and audio tracks. Tracks can start and end at any time. Each track
contains a stream of audio or video data.This spec mostly treats the formats used for stream audio and video data
as an implementation detail. In particular, whether stream buffers are compressed or uncompressed, what compression formats might be used, or what uncompressed formats might be used (e.g. audio sample rates, channels, and sample representation) are not specified, and are not directly observable. An implementation might even support changing formats over time within a single stream. Media data is implicitly resampled as necessary, e.g. when mixing streams with different formats. Non-normative suggestions for resampling algorithms will be provided in section 7.Built-in audio processing filters guarantee that if all the audio inputs constantly have the same uncompressed format
(same audio sample rate and channel configuration), the audio output will have the same format and there will be no unnecessary resampling.When samples are exposed to a Worker for processing, the user-agent chooses a fixed uncompressed audio format (sample rate and channel configuration) for its inputs and outputs; see section 4.4.
However, suggested resampling algorithms will be provided in an appendix.
partial interface MediaStream { readonly attribute double currentTime; ProcessedMediaStream createProcessor(optional in DOMString namedEffect); ProcessedMediaStream createWorkerProcessor(in Worker worker);};
ThecurrentTime
attribute returns the amount of time that thisMediaStream
has played since it was created.
ThecreateProcessor(namedEffect)
method returns a newProcessedMediaStream
with thisMediaStream
as its sole input.
ProcessedMediaStream
is configured with a built-in processing engine named bynamedEffect
, or the default processing engine ifnamedEffect
is omitted. IfnamedEffect
is not supported by this user-agent,createProcessor
returns null. User-agents adding nonstandard named effects should use vendor prefixing, e.g. "MozUnderwaterBubbles". The stream'sautofinish
attribute is set to true.ThecreateWorkerProcessor(worker)
method returns a newProcessedMediaStream
with thisMediaStream
as its sole input.
worker
as its processing engine. The stream'sautofinish
flag is set to true.Add event handlers or callback functions for all finished and blocking state changes?
We extend HTML media elements to produce and consume streams. When an HTML media element
produces a stream, it acts as a resource loader and control mechanism; the stream consists of whatever the media element is currently playing. When a media element consumes a stream, it acts a playback mechanism for the stream.partial interface HTMLMediaElement { readonly attribute MediaStream stream; MediaStream captureStream(); MediaStream captureStreamUntilEnded(); readonly attribute boolean audioCaptured; attribute any src;};
Thestream
attribute returns a stream which always plays whatever the element is playing. The stream is blocked while the media element is not playing. It is never finished, even when playback ends
stream
attribute for a given element always returns the same stream. When the stream changes to blocked, we fire thewaiting
event for the media element, and when it changes to unblocked we fire theplaying
event for the media element.Currently the HTML media element spec says thatplaying
would fire on an element that is able to play except that a downstreamMediaController
is blocked. This is incompatible with the above. I think that part of the HTML media spec should be changed so that only elements that are actually going to play fireplaying
.
ThecaptureStream()
method returns a newMediaStream
that plays the same audio and video asstream
.
captureStream()
sets theaudioCaptured
attribute to true.ThecaptureStreamUntilEnded()
method returns a newMediaStream
that plays the same audio and video asstream
, until the element next reaches
captureStreamUntilEnded()
sets theaudioCaptured
attribute to true.While the media element is playing a resource whose origin is not the same as the media element itself,MediaStream
s for the element do not play the content; they block.
This prevents leaking of media contents to scripts in the page. In the future we could relax this and allow the streams to play as long as there's no scripted processing of the data downstream, but that's trickier.
While theaudioCaptured
attribute is true, the element does not produce direct audio output.
stream
and the resource stream. This attribute is NOT reflected into the DOM. It is initially false. It's readonly to script and nothing ever sets it to true.Thesrc
attribute is extended to allow it to be set to aMediaStream
. The element
playbackRate
are not supported.TheURL.createObjectURL(stream)
method defined for HTML MediaStreams can create a URL to be
src
to the given stream.We add anAudio
constructor taking aMediaStream
as a parameter.
src
to the stream.[NamedConstructor=Audio(MediaStream src)]partial interface HTMLAudioElement {}
To enable precise control over the timing of attribute changes, many attributes can be set using a
"timed setter" method taking astartTime
parameter. The user-agent will attempt to make the change take effect when the subject stream's "current time" is exactly the givenstartTime
--- certainly no earlier, but possibly later if the change request is processed after the stream's current time has reachedstartTime
.startTime
is optional; if ommitted, the change takes effect as soon as possible.Using a timed setter method never changes the observed attribute value immediately. Setter method changes always take effect after the nextstable state, as described in section 2.1. Setting the attribute value changes the observed attribute value immediately, but the change to the underlying media stream will still not take effect until after
the next stable state.Multiple pending changes to an attribute are allowed. Calling the setter method with
startTime
T sets the value of the attribute for all times T' >= T to the desired value (wiping out the effects of previous calls to the setter method with a time greater than or equal tostartTime
). Therefore by calling the setter method multiple times with increasingstartTime
, a series of change requests can be built up. Setting the attribute directly sets the value of the attribute for all future times, wiping out any pending setter method requests.AProcessedMediaStream
combines zero or more input streams and applies some processing to
[Constructor(), Constructor(in Worker worker, in optional long audioSampleRate, in optional short audioChannels)]interface ProcessedMediaStream : MediaStream { readonly attribute MediaInput[] inputs; MediaInput addInput(in MediaStream inputStream, in optional double outputStartTime, in optional double inputStartTime); attribute any params; void setParams(in any params, in optional double startTime); readonly attribute boolean autofinish;};
The constructors create a newProcessedMediaStream
with no inputs.
ProcessedMediaStream
with a Worker processing engine, setting the audio sample rate toaudioSampleRate
and setting the number of audio channels toaudioChannels
(defaulting to 2). These parameters control the audio sample format used by the Worker (see below). Both constructors initializeautofinish
to false.Specify valid values foraudioChannels
andaudioSampleRate
.
Theinputs
attribute returns an array ofMediaInput
s, one for
ProcessedMediaStream
. (A stream can be used as multiple inputs to the sameProcessedMediaStream
.) It is initially empty if constructed via theProcessedMediaStream()
constructor, or contains a single element if constructed viaMediaStream.createProcessor
.AMediaInput
represents an input port. An input port isactive while it is enabled (see below) and the input stream is not finished.
TheaddInput(inputStream, outputStartTime, inputStartTime)
method adds a newMediaInput
to the end of the
inputs
array, whose input stream isinputStream
. TheoutputStartTime
andinputStartTime
parameters control when an input port is enabled and helps synchronize inputs with outputs. The input port is enabled when the input stream's current time isinputStartTime
and the output stream's current time isoutputStartTime
.More precisely, when theaddInput
call takes effect (see section 2.1),
outputStartTime
was omitted, set it to the output stream's current time.inputStartTime
was omitted, set it to the input stream's current time.inputStartTime
outputStartTime
inputStartTime
andoutputStartTime
. (This would be a good place for user-agents to emit a developer-accessible warning.)inputStartTime
, or the output stream's current time is less thanoutputStartTime
:inputStartTime
, block the input stream.outputStartTime
, block the output stream. var p = inputStream.createProcessor(); p.addInput(inputStream, 5);
In this example,inputStream
is used as an input top
twice.inputStream
must block untilp
has played 5s of output, but alsop
cannot play anything untilinputStream
unblocks. It seems hard to design an API that's hard to deadlock; even creating a cycle will cause deadlock.Theparams
attribute and thesetParams(params, startTime)
timed setter method set the paramters for this stream. On setting, astructured clone of this object is made. The clone is sent to
When an input stream finishes, at the next stable state anyMediaInput
s for that
When theautofinish
attribute is true, then when all stream inputs are finished (including if there are no inputs), the stream will automatically enter the finished state and never produce any more output (even if new inputs are attached).
AMediaInput
object controls how an input stream contributes to the combined stream.
interface MediaInput { readonly attribute MediaStream stream; MediaInput addFollowing(MediaStream inputStream); attribute double volume; void setVolume(in double volume, in optional double startTime, in optional double fadeTime); attribute any params; void setParams(in any params, in optional double startTime); attribute boolean blockInput; attribute boolean blockOutput; void remove(in optional double time);};
Thestream
attribute returns theMediaStream
connected to this input.
TheaddFollowing(inputStream)
method adds a newMediaInput
to
inputs
array for theProcessedMediaStream
thisMediaInput
belongs to. The newMediaInput
's input stream isinputStream
. The new input port is enabled when this input port's audio tracks all end, or the this input port is removed. Implementations should extrapolate and blend the ends of audio tracks if necessary to ensure a seamless transition between streams.Sequencing media resources using media element durations is possible, e.g.processor.addInput(video1.captureStreamUntilEnded()); processor.addInput(video2.captureStreamUntilEnded(),video1.duration);
. However, media element durations may not be perfectly reliable due to limitations of resource formats or commonly-used encoders. For example, WebM resources often have a slight mismatch between the number of audio samples and the duration metadata. For other resources, such as live streaming sources, the duration may not be knowable in advance. Thus, using the duration is often unreliable and additional API is warranted.
Add additional API to select particular tracks.
Thevolume
volume attribute and thesetVolume
timed setter method
setVolume
method takes an additionalfadeTime
parameter; when greater than zero, the volume is changed gradually from the value just beforestartTime
to the new value over the given fade time. The transition function is chosen so that if one stream changes from V1 to V2 and another stream changes from V2 to V1 over the same interval, the sum of the volumes at each point in time is V1 + V2. This attribute is initially 1.0.Specify the exact transition function. Tim says "w=cos((pi/2)*t)^2 for t=0...1".
Theparams
attribute and thesetParams(params, startTime)
timed setter method set the paramters for this input. On setting, astructured clone of this object is made. The clone is sent to
For the timed setter methods ofMediaInput
, the subject stream is the output stream, so changes take effect when the output stream's current time is equal tostartTime
.
TheblockInput
andblockOutput
attributes control
blockOutput
is true and the input port is active, if the input stream is blocked then the output stream must be blocked. While an active input is blocked and the output is not blocked, the input is treated as having no tracks. WhenblockInput
is true and the input port is active, if the output is blocked, then the input stream must be blocked. When false, while the output is blocked and an active input is not, the input will simply be discarded. These attributes are initially true.Need to look again at these. It's not clear we have use cases for both attributes, and I haven't implemented them yet and they could be hard to implement.
Theremove(time)
method removes thisMediaInput
from the inputs array of its owning
ProcessedMediaStream
at the given time relative to the output stream (or later, if it cannot be removed in time). Iftime
is omitted, the input is removed as soon as possible and theMediaInput
is removed from the destionation stream'sinput
array immediately. After removal, theMediaInput
object is no longer used; its attributes retain their current values and do not change unless explicitly set. All method calls are ignored. Additional calls toremove
with an earlier time can advance the removal time, but once removal is scheduled it cannot be stopped or delayed.AProcessedMediaStream
with a worker computes its output by dispatching a sequence ofonprocessmedia
callbacks to the worker, passing each aProcessMediaEvent
parameter. AProcessMediaEvent
provides audio sample buffers for each input stream. Each sample buffer for a givenProcessMediaEvent
has the same duration, so the inputs presented to the worker are always in sync. (Inputs may be added or removed betweenProcessMediaEvent
s, however.) The sequence of buffers provided for an input stream is the audio data to be played by that input stream. The user-agent will precompute data for the input streams as necessary.
For example, if a Worker computes the output sample for time T as a function of the [T - 1s, T + 1s] interval of an input stream, then initially the Worker would simply refuse to output anything until it has received at least 1s of input stream data, forcing the user-agent to precompute the input stream at least 1s ahead of the current time. (Note that large Worker latencies will increase the latency of changes to the media graph.)
Note thatWorker
s do not have access to most DOM API objects. In particular,Worker
s have no direct access toMediaStream
s.
Note that aProcessedMediaStream
's worker cannot be aSharedWorker
. This ensures that the worker can run in the same process as the page in multiprocess browsers, so media streams can be confined to a single process.
CurrentlyProcessMediaEvent
does not offer access to video data. This should be added later.
partial interface DedicatedWorkerGlobalScope { attribute Function onprocessmedia;};
Theonprocessmedia
attribute is the function to be called whenever stream data needs to be processed.
ProcessMediaEvent
is passed as the single parameter to each call to theonprocessmedia
callback. For a givenProcessedMediaStream
, the sameProcessMediaEvent
is passed in every call to theonprocessmedia
callback. This allows the callback function to maintain per-stream state.interface ProcessMediaEvent : Event { readonly attribute double inputTime; readonly attribute any params; readonly attribute double paramsStartTime; readonly attribute MediaInputBuffer inputs[]; readonly attribute long audioSampleRate; readonly attribute short audioChannels; reaodnly attribute long audioLength; void writeAudio(in Float32Array data); void finish();};
TheinputTime
attribute returns the duration of the input that has been consumed by the
ProcessedMediaStream
for this worker.Theparams
attribute provides a structured clone of the parameters object set by
ProcessedMediaStream.setParams
. The same object is returned in each event, except when the object has been changed bysetParams
between events.TheparamsStartTime
attribute returns the first time (measured in duration of input consumed for this stream) that thisparams
object was set.
Note that the parameters objects are constant over the duration of the inputs presented in the event. Frequent changes to parameters will reduce the length of the input buffers that can be presented to the worker.
inputs
provides access toMediaStreamBuffers
for each active input stream
ProcessedMediaStream.inputs
array).audioSampleRate
andaudioChannels
represent the format of the input and
audioSampleRate
is the number of samples per second.audioChannels
is the number of channels; the channel mapping is as defined in the Vorbis specification. These values are constant for a givenProcessedMediaStream
. When theProcessedMediaStream
was constructed using the Worker constructor, these values are the values passed as parameters there. When theProcessedMediaStream
was constructed viaMediaStream.createProcessor
, the values are chosen to match the first active input stream (or 44.1KHz, 2 channels if there is no active input stream).audioLength
is the duration of the input(s) multiplied by the sample rate. If there are no inputs,
writeAudio(data)
writes audio data to the stream output.
kind
is "main" and the other metadata attriutes are the empty string. The data for the output audio track is the concatenation of the inputs to eachwriteAudio
call before the event handler returns. The data buffer is laid out with the channels non-interleaved, as for the input buffers (see below). The length ofdata
must be a multiple ofaudioChannels
; if not, then only the sample values up to the largest multiple ofaudioChannels
less than the data length are used.It is permitted to write less audio than the duration of the inputs (including none). This indicates latency in the filter. Normally the user-agent will dispatch another event to provide
more input until the worker starts producing output. It is also permitted to write more audio than the duration of the inputs, for example if there are no inputs. Filters with latency should respond to an event with no inputs by writing out some of their buffered data; the user-agent is draining them.A synthesizer with no inputs can output as much data as it wants; the UA will buffer data and fire events as necessary. Filters that misbehave, e.g. by always writing zero-length buffers, will cause the stream to block due to an underrun.
IfwriteAudio
is not called during the event handler, then the input audio buffers are added together and written to the output.
IfwriteAudio
is called outside the event handler, the call is ignored.
Callingfinish()
puts the stream into the finished state (once any previously buffered output has been consumed). The event callback will never be called again.finish()
can be called at any time, inside or outside the event handler.
The output video track is computed as if there was no worker (see above).
This will change when we add video processing.
interface MediaInputBuffer { readonly attribute any params; readonly attribute double paramsStartTime; readonly attribute Float32Array audioSamples;};
Theparams
attribute provides a structured clone of the parameters object set by
MediaInput.setParams
. The same object is returned in each event, except when the object has been changed bysetParams
between events.TheparamsStartTime
attribute returns the first time (measured in duration of input consumed for this stream) that thisparams
object was set.
audioSamples
gives access to the audio samples for each input stream. The array length will beevent.audioLength
multiplied byevent.audioChannels
. The samples are floats ranging from -1 to 1, laid out non-interleaved, i.e. consecutive segments ofaudioLength
samples each. The durations of the input buffers for the input streams will be equal. TheaudioSamples
object will be a fresh object in each event. For inputs with no audio track,audioSamples
will be all zeroes.
AProcessedMediaStream
with the default processing engine produces output as follows:
id
,kind
,label
, andlanguage
) is equal to that of the audio track for the last active input that has an audio track. The output audio track is produced by adding the samples of the audio tracks of the active inputs together.id
,kind
,label
, andlanguage
) is equal to that of the video track for the last active input that has a video track. The output video track is produced by compositing together all the video frames from the video tracks of the active inputs, with the video frames from higher-numbered inputs on top of the video frames from lower-numbered inputs; each video frame is letterboxed to the size of the video frame for the last active input that has a video track.This means if the last input's video track is opaque, the video output is simply the video track of the last input.
AProcessedMediaStream
with the "LastInput" processing engine simply produces the last active input stream as output. If there are no active input streams, it produces the same output as the default processing engine.
While aProcessedMediaStream
has itself as a direct or indirect input stream (considering only active inputs), it is blocked.
At each moment, every stream should not be blocked except as explicitly required by this specification.
To enable video synthesis and some easy kinds of video effects we can record the contents of a canvas:
partial interface HTMLCanvasElement { readonly attribute MediaStream stream;};
Thestream
attribute is a stream containing a video track with the "live" contents of the canvas as video frames whose size is the size of the canvas, and no audio track. It always returns the same stream for a given element.
Here will be some non-normative implementation suggestions.
Add Worker scripts for these examples.
<video src="foo.webm" controls></video><audio autoplay></audio><script>document.getElementById("out").src = document.getElementById("v").captureStream().createWorkerProcessor(new Worker("effect.js"));</script>
<video src="foo.webm"></video><audio src="back.webm"></audio><audio autoplay></audio><script> var mixer = document.getElementById("v").captureStream().createWorkerProcessor(new Worker("audio-ducking.js")); mixer.addInput(document.getElementById("back").captureStream()); document.getElementById("out").src = mixer; function startPlaying() { document.getElementById("v").play(); document.getElementById("back").play(); } // MediaController is a more convenient API because it ties together control of the elements, // but using streams is more flexible (e.g. they can be seeked to different offsets).</script>
<script> navigator.getUserMedia('audio', gotAudio); function gotAudio(stream) { peerConnection.addStream(stream.createWorkerProcessor(new Worker("effect.js"))); }</script>
<canvas></canvas><script> navigator.getUserMedia('audio', gotAudio); var streamRecorder; function gotAudio(stream) { var worker = new Worker("visualizer.js"); var processed = stream.createWorkerProcessor(worker); worker.onmessage = function(event) { drawSpectrumToCanvas(event.data, document.getElementById("c")); } streamRecorder = processed.record(); peerConnection.addStream(processed); }</script>
<canvas></canvas><audio src="back.webm"></audio><script> navigator.getUserMedia('audio', gotAudio); var streamRecorder; function gotAudio(stream) { var worker = new Worker("visualizer.js"); var processed = stream.createWorkerProcessor(worker); worker.onmessage = function(event) { drawSpectrumToCanvas(event.data, document.getElementById("c")); } var mixer = processed.createProcessor(); mixer.addInput(document.getElementById("back").captureStream()); streamRecorder = mixer.record(); peerConnection.addStream(mixer); }</script>
<script> var worker = new Worker("spatializer.js"); var spatialized = stream.createWorkerProcessor(worker); peerConnection.onaddstream = function (event) { spatialized.addInput(event.stream).params = {x:..., y:..., z:...}; }; (new Audio(spatialized)).play();</script>
This method requires that you know each stream's duration, which is a bit suboptimal. To get around that we'd need new API, perhaps a new kind of ProcessedMediaStream that plays streams in serial.
<audio src="in1.webm" preload></audio><audio src="in2.webm"></audio><script> var in1 = document.getElementById("in1"); in1.onloadedmetadata = function() { var mixer = in1.captureStreamUntilEnded().createProcessor(); var in2 = document.getElementById("in2"); mixer.addInput(in2.captureStreamUntilEnded(), in1.duration); (new Audio(mixer)).play(); in1.play(); }</script>
There are two ways to implement seamless switching: seek the second resource to before the current time and then run the decoder faster than real-time to catch up to the first resource's play point, or seek the second resource to after the current time and enable it when the first resource catches up to the seek point. The first is more robust if the seek takes unexpectedly long, but the second is less demanding on the decoder. Only the second method is currently implementable with this API (since by design there is no way to drive MediaStreams faster than real-time). If we want to support the first method as well, the right way would be to add API to media elements to let them seek to synchronize with a given MediaStream.
<audio src="in1.webm" preload></audio><audio src="in2.webm"></audio><audio autoplay></audio><script> var stream1 = document.getElementById("in1").captureStream(); var mixer = stream1.createProcessor("LastInput"); document.getElementById("out").src = mixer; function switchStreams() { var in2 = document.getElementById("in2"); in2.currentTime = in1.currentTime + 10; // arbitrary, but we should be able to complete the seek within this time mixer.addInput(in2.captureStream(), mixer.currentTime + 10); in2.play(); // in2 will be blocked until the input port is enabled in2.onplaying = function() { mixer.inputs[0].remove(); }; }</script>
<audio autoplay></audio><script> document.getElementById("out").src = new ProcessedMediaStream(new Worker("synthesizer.js"));</script>
<script> var effectsMixer = ...; function playSound(src) { var audio = new Audio(src); audio.oncanplaythrough = function() { var stream = audio.captureStreamUntilEnded(); var port = effectsMixer.addInput(stream); port.blockOutput = false; audio.play(); } }</script>
<script> var effectsMixer = ...; var audio = new Audio(...); function triggerSound() { var sound = audio.clone(); var stream = sound.captureStreamUntilEnded(); sound.play(); effectsMixer.addInput(stream, effectsMixer.currentTime + 5); }</script>
<script> navigator.getUserMedia('video', gotVideo); function gotVideo(stream) { stream.createWorkerProcessor(new Worker("face-recognizer.js")); }</script>
<script> navigator.getUserMedia('video', gotVideo); var streamRecorder; function gotVideo(stream) { streamRecorder = stream.record(); } function stopRecording() { streamRecorder.getRecordedData(gotData); } function gotData(blob) { var x = new XMLHttpRequest(); x.open('POST', 'uploadMessage'); x.send(blob); }</script>
<canvas width="640" height="480"></canvas><script> var canvas = document.getElementById("c"); var streamRecorder = canvas.stream.record(); function stopRecording() { streamRecorder.getRecordedData(gotData); } function gotData(blob) { var x = new XMLHttpRequest(); x.open('POST', 'uploadMessage'); x.send(blob); } var frame = 0; function updateCanvas() { var ctx = canvas.getContext("2d"); ctx.clearRect(0, 0, 640, 480); ctx.fillText("Frame " + frame, 0, 200); ++frame; } setInterval(updateCanvas, 30);</script>