This document is a draft specification proposal with no official status. Send comments to the W3C audio mailing list, or Robert O'Callahan. It is inappropriate to cite this document except as a work in progress.
A number of existing or proposed features for the Web platform deal with continuous real-time media:
The ideas here build on Ian Hickson's proposal for HTML Streams, adding features partly inspired by the Mozilla audio API and the 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.
MediaStreams and MediaInputs refer to the DOM-visible state, and references to media streams and input 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 be blocked, 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 the finished 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 and ProcessedMediaStreams (see below). This avoids situations where streams that aren't connected
to anything and could be garbage-collected have to be kept running because their stopping could possibly be observed (e.g. because a Worker is being used to generate data for the stream and the Worker can observe whether the 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);
};The currentTime attribute returns the amount of time that this MediaStream has played since it was created.
The createProcessor(namedEffect) method returns a new ProcessedMediaStream with this MediaStream as its sole input.
The ProcessedMediaStream is configured with a built-in processing engine named by namedEffect,
or the default processing engine if namedEffect is omitted. If namedEffect 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's autofinish attribute
is set to true.
The createWorkerProcessor(worker) method returns a new ProcessedMediaStream with this MediaStream as its sole input.
The stream is configured with worker as its processing engine.
The stream's autofinish 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;
};The stream 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
(since the element might load a new resource or seek within the current resource). The stream attribute for a given element always returns
the same stream. When the stream changes to blocked, we fire the waiting event for the media element,
and when it changes to unblocked we fire the playing event for the media element.
Currently the HTML media element spec says that playing would fire on an element
that is able to play except that a downstream MediaController 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 fire playing.
The captureStream() method returns a new MediaStream that plays the same audio and video as stream.
captureStream() sets the audioCaptured attribute to true.
The captureStreamUntilEnded() method returns a new MediaStream that plays the same audio and video as stream, until the element next reaches
the "ended playback" state, at which point this stream will enter the finished state.
captureStreamUntilEnded() sets the audioCaptured attribute to true.
While the media element is playing a resource whose origin is not the same as the media element itself, MediaStreams 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 the audioCaptured attribute is true, the element does not produce direct audio output.
Audio output is still sent to 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.
The src attribute is extended to allow it to be set to a MediaStream. The element
will play the contents of the given stream. This is treated as a live stream; seeking and playbackRate
are not supported.
The URL.createObjectURL(stream) method defined for HTML MediaStreams can create a URL to be
used as a source for a media element. Setting a media element to use this source URL is equivalent to setting
the media element src to the given stream.
We add an Audio constructor taking a MediaStream as a parameter.
This sets the initial 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 a startTime parameter. The user-agent will attempt to make the change take
effect when the subject stream's "current time" is exactly the given startTime --- certainly no earlier, but possibly later if the change request is processed after the stream's current time has reached startTime. 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 next stable 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 to startTime). Therefore
by calling the setter method multiple times with increasing startTime, 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.
A ProcessedMediaStream combines zero or more input streams and applies some processing to
combine them into a single output stream.
[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 new ProcessedMediaStream with no inputs.
The second constructor creates a new ProcessedMediaStream with a Worker
processing engine, setting the
audio sample rate to audioSampleRate and setting the number of
audio channels to audioChannels (defaulting to 2). These parameters control the audio sample
format used by the Worker (see below). Both constructors initialize autofinish to false.
Specify valid values for audioChannels and audioSampleRate.
The inputs attribute returns an array of MediaInputs, one for
each stream currently configured as an input to the ProcessedMediaStream. (A stream can be used as multiple inputs to the same ProcessedMediaStream.) It is
initially empty if constructed via the ProcessedMediaStream() constructor, or
contains a single element if constructed via MediaStream.createProcessor.
A MediaInput represents an input port. An input port is active while it is enabled (see below) and the input stream is not finished.
The addInput(inputStream, outputStartTime, inputStartTime) method adds a new MediaInput to the end of the
inputs array, whose input stream is inputStream. The outputStartTime
and inputStartTime 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 is
inputStartTime and the output stream's current time is outputStartTime.
More precisely, when the addInput call takes effect (see section 2.1),
the user-agent runs the following steps:
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 and outputStartTime. (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 than outputStartTime:
inputStartTime, block the input stream.
outputStartTime, block the output stream.
var p = inputStream.createProcessor();
p.addInput(inputStream, 5);inputStream is used as an input to p twice. inputStream must block until p has played 5s of output, but also p cannot play anything until inputStream unblocks. It seems hard to design an API that's hard to deadlock; even creating a cycle will cause deadlock.The params attribute and the setParams(params, startTime) timed setter method set the paramters for this stream. On setting, a structured clone of this object is made. The clone is sent to
the worker (if there is one) during media processing. On getting, a fresh clone is returned.
When an input stream finishes, at the next stable state any MediaInputs for that
stream are automatically removed.
When the autofinish 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).
A MediaInput 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);
};The stream attribute returns the MediaStream connected to this input.
The input stream is treated as having at most one audio and/or video track; all enabled audio tracks are mixed
together and the rest are dropped, and all video tracks other than the selected video track are dropped.
The addFollowing(inputStream) method adds a new MediaInput to
the end of the inputs array for the ProcessedMediaStream this MediaInput
belongs to. The new MediaInput's input stream is inputStream. 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.
The volume volume attribute and the setVolume timed setter method
control the input volume; the input stream's audio is multiplied by this volume before
being processed. The setVolume method takes an additional fadeTime parameter; when greater
than zero, the volume is changed gradually from the value just before startTime 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".
The params attribute and the setParams(params, startTime) timed setter method set the paramters for this input. On setting, a structured clone of this object is made. The clone is sent to
the worker (if there is one) during media processing. On getting, a fresh clone is returned.
For the timed setter methods of MediaInput, the subject stream is the output stream, so changes take effect when the output stream's current time is equal to startTime.
The blockInput and blockOutput attributes control
how the blocking status of the input stream is related to the blocking status of the output stream.
When 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. When blockInput 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.
The remove(time) method removes this MediaInput 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).
If time is omitted, the input is removed as soon as possible and the MediaInput is removed from
the destionation stream's input array immediately.
After removal, the MediaInput 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 to remove with an earlier time
can advance the removal time, but once removal is scheduled it cannot be stopped or delayed.
A ProcessedMediaStream with a worker computes its output by dispatching a sequence of onprocessmedia callbacks to the worker, passing each a ProcessMediaEvent parameter. A ProcessMediaEvent provides audio sample buffers for each input stream. Each sample buffer for a given ProcessMediaEvent has the same duration, so the inputs presented to the worker are always in sync. (Inputs may be added or removed between ProcessMediaEvents, 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 that Workers do not have access to most DOM API objects. In particular, Workers have no direct access to MediaStreams.
Note that a ProcessedMediaStream's worker cannot be a
SharedWorker. 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.
Currently ProcessMediaEvent does not offer access to video data. This should be added later.
partial interface DedicatedWorkerGlobalScope {
attribute Function onprocessmedia;
};The onprocessmedia attribute is the function to be called whenever stream data needs to be processed.
A ProcessMediaEvent is passed as the single parameter to each call to the onprocessmedia callback.
For a given ProcessedMediaStream, the same ProcessMediaEvent is passed in every call to the
onprocessmedia 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();
};The inputTime attribute returns the duration of the input that has been consumed by the
ProcessedMediaStream for this worker.
The params 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 by setParams between events.
The paramsStartTime attribute returns the first time (measured in duration of input consumed for this stream) that this params 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 to MediaStreamBuffers for each active input stream
(in the same order as those streams appear in the ProcessedMediaStream.inputs array).
audioSampleRate and audioChannels represent the format of the input and
output audio sample buffers. 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 given ProcessedMediaStream. When the ProcessedMediaStream
was constructed using the Worker constructor, these values are the values passed as parameters there. When the
ProcessedMediaStream was constructed via MediaStream.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,
the user-agent will choose a value representing the suggested amount of audio that the worker should produce.
writeAudio(data) writes audio data to the stream output.
The output has a single audio track. If there is an active input with an audio track, then the metadata for the output audio track is set to the metadata for the audio track of the last active input that has an audio track, otherwise the output audio track's 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 each writeAudio 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 of data must be
a multiple of audioChannels; if not, then only the sample values up to the largest multiple of audioChannels 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.
If writeAudio is not called during the event handler, then the input audio buffers are added together and written to the output.
If writeAudio is called outside the event handler, the call is ignored.
Calling finish() 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;
};The params 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 by setParams between events.
The paramsStartTime attribute returns the first time (measured in duration of input consumed for this stream) that this params object was set.
audioSamples gives access to the audio samples for each input stream. The array length will be event.audioLength multiplied by event.audioChannels. The samples are floats ranging from -1 to 1, laid out non-interleaved, i.e. consecutive segments of audioLength samples each. The durations of the input buffers for the input streams will be equal. The audioSamples object will be a fresh object in each event. For inputs with no audio track, audioSamples will be all zeroes.
A ProcessedMediaStream with the default processing engine produces output as follows:
id, kind, label, and language)
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, and language)
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.
A ProcessedMediaStream 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 a ProcessedMediaStream 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;
};The stream 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" id="v" controls></video>
<audio id="out" autoplay></audio>
<script>
document.getElementById("out").src =
document.getElementById("v").captureStream().createWorkerProcessor(new Worker("effect.js"));
</script><video src="foo.webm" id="v"></video>
<audio src="back.webm" id="back"></audio>
<audio id="out" 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 id="c"></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 id="c"></canvas>
<audio src="back.webm" id="back"></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" id="in1" preload></audio>
<audio src="in2.webm" id="in2"></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" id="in1" preload></audio>
<audio src="in2.webm" id="in2"></audio>
<audio id="out" 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 id="out" 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" id="c"></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>